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The Ear - DSD under Fire


Nikhil

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Yes but as the resolution increases these differences diminish and, given optimal implementation, actually vanish.

 

Sounds logical, but I disagree in the practical implementation. As long as format conversions are required in producing, delivering, and playing recorded music, the negative affects of those conversions take a sound quality toll. The fact is all A/D converters in use today, and the vast majority of D/A converters are front and back ended with multi megahertz Sigma Delta modulators, and produce or generate multi megahertz bit streams. Converting the input and output format for the sake of post processing and/or the delivery medium is not the path I and others in the trade favor. It's finishing the tools development started by Sony that allow end to end production and delivery in one format that's PDM.

 

Define it away with esoteric theoretical discussions to your hearts content, but that's the reality today. You're not getting all the sound quality your money is paying for. More than you realize in most productions remains on the proverbial cutting room floor.

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Different world views, I guess - I would think of draining the swamp as the mechanism that performs the function of keeping one's ass away from alligators. Oh well, so much for communication between an engineer and a lawyer. ;)

 

Whereas I would simply hire a boatload or two of Cajuns. I bet I could get em cheap too, with the promise of good hunting and new purses and shoes for their ladies...

 

End result, swamp with no gators, and no environmental protection lawsuits. :)

 

-Paul

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

Robert A. Heinlein

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Yes but as the resolution increases these differences diminish and, given optimal implementation, actually vanish. The point at which the differences vanish is the point at which increasing the sampling frequency has no further benefit. I will leave the actual mathematical proof of this to the reader :)

 

You may be on to something here. Although I don't get the math I do prefer 16/44.1 PCM upsampled to 384KHz on my setup even with HQPlayer. I have to admit that my setup is older and possibly tuned towards PCM by design as Paul R mentioned earlier. PCM upsampled to DSD doesn't sound as good.

 

Native DSD is another discussion altogether. But even when downsampled to PCM the sound quality is excellent.

 

I mention setup design because when my amps were built 7-8 years ago RBCD was it. My preamp's tube design intent was to add a little warmth to the dry sound of RBCD. With the advent of hi res audio my amp manufacturer has moved to a SS design altogether. I just wonder how all this ties in with sound reproduction. I do plan to try out a SS design preamp in my setup.

Custom Win10 Server | Mutec MC-3+ USB | Lampizator Amber | Job INT | ATC SCM20PSL + JL Audio E-Sub e110

 

 

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Not really responding to anyone's comments in particular here...

 

At the end of the day, when discussing DSD vs PCM we have to distinguish different situations.

 

When recording (which is something as listeners we have no control over), and I'm talking in the real world with real equipment readily available, there is a choice between the output of your ADC being DSD or PCM. Now its more a preference if you prefer the sound quality of DSD or PCM, but the truth is that if you are going to do much in the way of editing you are going to be faced with the reality that your "pure" DSD is going to be converted to PCM / DXD for editing. For that reason it doesn't make much sense to record in DSD if you are going to do much editing (IMO).

 

Now its a mathematical fact that any conversion DSD to PCM is inherently lossy. That isn't to say anything about the quality of one vs the others; lossy is simply a technical term which says that once you convert one way, there is no way to cover BACK to get EXACTLY the original - you only ever get approximately there.

 

Such mathematics can be seen in everyday arithmetic. (Ignoring what your calculator might tell you with such simple maths) Multiplication by 1/3 is a example of a lossy function. That is if you take 1 and multiply it by a third in a calculator you get 0.3333333. Now take 0.3333333 and multiply it by 3 - you will get .9999999 (some calculators will round this up).

 

Now once you do any editing, what are you going to do? You have a DXD file. Does it really seem right to then convert that back to DSD. What is going to happen further down the line when it comes to mastering?

 

So then we move on to the DAC which the listener has at home. (Almost) Everyone these days has the ability to play both PCM and DSD files - applications such as J.River have supported this for many years now; so what format the music is available in, and what format your DAC supports is close to irrelevant in terms of what you can play.

 

If your DAC sounds better if fed DSD, then convert your PCM to DSD. If your DAC sounds better or doesn't support PCM do the opposite. So wouldn't you (as an audiophile) prefer to have that "music" in as close to its original native format as possible? There is nothing inherently superior about the conversion available to studios these days...

 

So to summarise, DSD as a recording format only makes sense when recording something which is not going to be further edited. Everything else we want in a format as close to the output of the editing suite as practical.

 

Now perhaps there does need to be something else though ... a development which will allow the SDM to be passed to the producers console rather than the requirement for DSD or PCM. As I understand it this is something Miska works towards (but at the other end of the chain) but while theoretically interesting is not of practical use in the real world while its a Miska only project.

 

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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Eloise - what about avoiding DSD to DXD/PCM and PCM to DSD conversion in realtime with limited hardware level? Can offline (=software) conversion do the job better than (in some cases even ancient :)) hardware chip with limited computing capabilities?

Sorry, english is not my native language.

Fools and fanatics are always certain of themselves, but wiser people are full of doubts.

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So to summarise, DSD as a recording format only makes sense when recording something which is not going to be further edited.

 

There are several production workflows in which DSD can be used:

 

- acoustic recordings made direct to DSD

- acoustic recordings with on-site analog mixing & native DSD editing

- direct transfers from analog tapes

- all projects that use analog mixing consoles

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I mention setup design because when my amps were built 7-8 years ago RBCD was it. My preamp's tube design intent was to add a little warmth to the dry sound of RBCD. With the advent of hi res audio my amp manufacturer has moved to a SS design altogether. I just wonder how all this ties in with sound reproduction. I do plan to try out a SS design preamp in my setup.

 

Many SS amps still sound very analog, see Heed Audio amps for example. So if you want to hear how PCM DIGITAL really sounds, your best bet is with a class D amplifier.

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Of course if one is using for example digital synthesizers during production, with:

 

"Aliasing – they all have some amount of nasty harmonic distortion caused by sample aliasing. Some do reduce this to a minimum bit it’s always there.

 

Scratchy transients – they just can’t seem to handle situations involving high/short envelope settings in which you get nasty split, crackly distortion in onset transients. You often want a spike at the onset of a sound to give it a snap that your ear can catch hold of, so you might have a short envelope opening a filter momentarily, but this often introduces some scratchy high frequencies or crackly transients.

 

Gritty high end – By far my most serious complaint. The thing that always puts me off. The high frequency component in every digital synth I have ever used always has a kind of brittle, rough texture. The sound is like subtle bit-depth reduction all over the top end. Digital synths never fizz cleanly and I always find it distracting.

 

Bad aliasing during even subtle pitch modulation – one of the main characteristics of analogue synths is looseness. Pitched oscillators will slide around, if only very slightly. Many will recreate this via slight pitch modulation, say settings an LFO to wobble oscillator pitch or using some pitch shift plug-in, but these, even very subtle, variations immediately incur aliasing with a lot of digital synths.

 

Break down at all extremes – digital synths always break apart with extreme settings. You can’t have LFOs going too fast, fast attack envelopes crackle and spit, high resonance sounds aliased, high register notes sound scratchy.

 

Unstable low end – I have noticed that, to my memory, every digital synth I have used unravels on very low frequencies depending on settings. The low end seems to be detached from the rest of the signal. Maybe something to do with phase coherence of digital filters. Dunno."

 

https://marcdhall.wordpress.com/2014/05/18/synths-analogue-versus-digital/

 

they are free to keep that recording in PCM...

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There are several production workflows in which DSD can be used:

 

- acoustic recordings made direct to DSD

- acoustic recordings with on-site analog mixing & native DSD editing

- direct transfers from analog tapes

- all projects that use analog mixing consoles

Hiro... Those would be production work flows where there is no further editing.

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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Eloise - what about avoiding DSD to DXD/PCM and PCM to DSD conversion in realtime with limited hardware level? Can offline (=software) conversion do the job better than (in some cases even ancient :)) hardware chip with limited computing capabilities?

Yes I agree that off line conversion DSD to PCM (or vice versa) may be preferable; but that could be done by the end listener rather than it being dictated by the producer / mastering / record label.

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.

Link to comment
Not really responding to anyone's comments in particular here...

 

At the end of the day, when discussing DSD vs PCM we have to distinguish different situations.

 

When recording (which is something as listeners we have no control over), and I'm talking in the real world with real equipment readily available, there is a choice between the output of your ADC being DSD or PCM. Now its more a preference if you prefer the sound quality of DSD or PCM, but the truth is that if you are going to do much in the way of editing you are going to be faced with the reality that your "pure" DSD is going to be converted to PCM / DXD for editing. For that reason it doesn't make much sense to record in DSD if you are going to do much editing (IMO).

 

Now its a mathematical fact that any conversion DSD to PCM is inherently lossy. That isn't to say anything about the quality of one vs the others; lossy is simply a technical term which says that once you convert one way, there is no way to cover BACK to get EXACTLY the original - you only ever get approximately there.

 

Such mathematics can be seen in everyday arithmetic. (Ignoring what your calculator might tell you with such simple maths) Multiplication by 1/3 is a example of a lossy function. That is if you take 1 and multiply it by a third in a calculator you get 0.3333333. Now take 0.3333333 and multiply it by 3 - you will get .9999999 (some calculators will round this up).

 

Now once you do any editing, what are you going to do? You have a DXD file. Does it really seem right to then convert that back to DSD. What is going to happen further down the line when it comes to mastering?

 

So then we move on to the DAC which the listener has at home. (Almost) Everyone these days has the ability to play both PCM and DSD files - applications such as J.River have supported this for many years now; so what format the music is available in, and what format your DAC supports is close to irrelevant in terms of what you can play.

 

If your DAC sounds better if fed DSD, then convert your PCM to DSD. If your DAC sounds better or doesn't support PCM do the opposite. So wouldn't you (as an audiophile) prefer to have that "music" in as close to its original native format as possible? There is nothing inherently superior about the conversion available to studios these days...

 

So to summarise, DSD as a recording format only makes sense when recording something which is not going to be further edited. Everything else we want in a format as close to the output of the editing suite as practical.

 

Now perhaps there does need to be something else though ... a development which will allow the SDM to be passed to the producers console rather than the requirement for DSD or PCM. As I understand it this is something Miska works towards (but at the other end of the chain) but while theoretically interesting is not of practical use in the real world while its a Miska only project.

 

Eloise

 

Hi Eloise,

 

Agree in principle. I think if I'm understanding correctly what people like tailspn are saying, the DSD/PCM conversion for editing can be done just to the portion of the file that must be edited. If that's the case, then presumably one wouldn't necessarily want to convert an entire file to PCM for listening because a few tenths of seconds had to be converted for editing. (If that's not the case, forget I said anything. :) )

 

One other addition: Regarding the "choice of output" of an ADC being DSD or PCM, I believe the "native" output of all studio ADCs in common use would be DSD, but that it is most commonly converted to PCM in the ADC (pretty analogous to the situation with most consumer DACs).

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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As an owner of a large amount of synths I can't argue with any of what you say...

 

... the problem is that no digital synth is remotely built to audiophile standards (containing cheap DACs and components) and those using wave samples are usually samples that are compressed to death to save memory.

 

I think 'virtual analog synthesis' still has a long way to go but we should eventually get somewhere near perfection. 'Uhe Diva' is a nice virtual analogue, not perfect but very nice.

 

Of course plenty of analog synths have their issues too. It's nice to mix and match analog and digital to counter each's pros and cons... just like great teamwork.

 

;-)

 

 

Of course if one is using for example digital synthesizers during production, with:

 

"Aliasing – they all have some amount of nasty harmonic distortion caused by sample aliasing. Some do reduce this to a minimum bit it’s always there.

 

Scratchy transients – they just can’t seem to handle situations involving high/short envelope settings in which you get nasty split, crackly distortion in onset transients. You often want a spike at the onset of a sound to give it a snap that your ear can catch hold of, so you might have a short envelope opening a filter momentarily, but this often introduces some scratchy high frequencies or crackly transients.

 

Gritty high end – By far my most serious complaint. The thing that always puts me off. The high frequency component in every digital synth I have ever used always has a kind of brittle, rough texture. The sound is like subtle bit-depth reduction all over the top end. Digital synths never fizz cleanly and I always find it distracting.

 

Bad aliasing during even subtle pitch modulation – one of the main characteristics of analogue synths is looseness. Pitched oscillators will slide around, if only very slightly. Many will recreate this via slight pitch modulation, say settings an LFO to wobble oscillator pitch or using some pitch shift plug-in, but these, even very subtle, variations immediately incur aliasing with a lot of digital synths.

 

Break down at all extremes – digital synths always break apart with extreme settings. You can’t have LFOs going too fast, fast attack envelopes crackle and spit, high resonance sounds aliased, high register notes sound scratchy.

 

Unstable low end – I have noticed that, to my memory, every digital synth I have used unravels on very low frequencies depending on settings. The low end seems to be detached from the rest of the signal. Maybe something to do with phase coherence of digital filters. Dunno."

 

https://marcdhall.wordpress.com/2014/05/18/synths-analogue-versus-digital/

 

they are free to keep that recording in PCM...

Source:

*Aurender N100 (no internal disk : LAN optically isolated via FMC with *LPS) > DIY 5cm USB link (5v rail removed / ground lift switch - split for *LPS) > Intona Industrial (injected *LPS / internally shielded with copper tape) > DIY 5cm USB link (5v rail removed / ground lift switch) > W4S Recovery (*LPS) > DIY 2cm USB adaptor (5v rail removed / ground lift switch) > *Auralic VEGA (EXACT : balanced)

 

Control:

*Jeff Rowland CAPRI S2 (balanced)

 

Playback:

2 x Revel B15a subs (balanced) > ATC SCM 50 ASL (balanced - 80Hz HPF from subs)

 

Misc:

*Via Power Inspired AG1500 AC Regenerator

LPS: 3 x Swagman Lab Audiophile Signature Edition (W4S, Intona & FMC)

Storage: QNAP TS-253Pro 2x 3Tb, 8Gb RAM

Cables: DIY heavy gauge solid silver (balanced)

Mains: dedicated distribution board with 5 x 2 socket ring mains, all mains cables: Mark Grant Black Series DSP 2.5 Dual Screen

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Well' date=' I've listened to a counterpointing school of thought from Marantz' Ken Ishiwata, to paraphrase him (for I've not time right now to locate and transcribe verbatim) « number of bits are not that important, what's important is the sampling frequency. »

 

There is some truth to that statement, though it's far from the whole truth. Given a band-limited analogue signal, in order to represent it digitally, we absolutely must use a sampling rate of more than twice the highest frequency component in the signal (Nyquist). If we fail here, no amount of bits per sample can make up for it. Beyond this absolute threshold, the noise level of the digital representation at a given sampling rate decreases as the number of bits per sample increases. To achieve a certain noise level, the number of bits per sample required goes down as the sampling rate goes up. Once a bit of headroom has been established, there is something of a duality between bit depth and sampling rate allowing us to trade one for the other.

 

I recall that he also advocates towards « variable bitrate »

 

Using a variable bitrate never produces a more accurate result than would keeping it constant at the peak level, although it can give equal quality using fewer bits. Most compression schemes, e.g. FLAC, have a variable bitrate for this reason.

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There are several production workflows in which DSD can be used:

 

- acoustic recordings made direct to DSD

- acoustic recordings with on-site analog mixing & native DSD editing

- direct transfers from analog tapes

- all projects that use analog mixing consoles

 

Do you have a keyboard macro to post that reply?

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As an owner of a large amount of synths I can't argue with any of what you say...

 

... the problem is that no digital synth is remotely built to audiophile standards (containing cheap DACs and components) and those using wave samples are usually samples that are compressed to death to save memory.

 

I think 'virtual analog synthesis' still has a long way to go but we should eventually get somewhere near perfection. 'Uhe Diva' is a nice virtual analogue, not perfect but very nice.

 

Of course plenty of analog synths have their issues too. It's nice to mix and match analog and digital to counter each's pros and cons... just like great teamwork.

 

;-)

 

Some analogue synthesisers are very hard to recreate digitally, e.g. if they use non-linear components, and I see no reason why we should attempt it. After all, we still use analogue instruments like pianos and guitars (both acoustic and electric). On the flip side, some digital effects would be very hard to create using only analogue components.

 

If using a digital synthesiser to produce a recording, it ought to be possible to feed the digital signal directly into a DAW rather than going via an analogue path, in which case cheap DACs wouldn't be an issue. Maybe this is how it's done; I'm not involved in music production.

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Agree in principle. I think if I'm understanding correctly what people like tailspn are saying, the DSD/PCM conversion for editing can be done just to the portion of the file that must be edited. If that's the case, then presumably one wouldn't necessarily want to convert an entire file to PCM for listening because a few tenths of seconds had to be converted for editing. (If that's not the case, forget I said anything. )

I think you're right in that only the portion being edited is converted.

 

One other addition: Regarding the "choice of output" of an ADC being DSD or PCM, I believe the "native" output of all studio ADCs in common use would be DSD, but that it is most commonly converted to PCM in the ADC (pretty analogous to the situation with most consumer DACs).

Perhaps I wasn't clear ... but I was considering the "output" of the ADC as a recording unit (perhaps I should have said a recording interface rather than a ADC) - that is if you feed a microphone into the input of an ADC, the "output" via a USB or other digital interface is either PCM or DSD. For most ADC / recording interfaces that is going to be PCM regardless of what happens "inside the box".

 

Most people are going to have no influence over what happens "inside the box" so (to me) arguing about what is happening inside the box is irrelevant.

 

As I read tailspn correctly, what is needed is recording interfaces which will output the SDM, which can then be edited without conversion. There would only need to be a conversion at the Mastering stage to the chosen DSD or PCM...

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.

Link to comment
As an owner of a large amount of synths I can't argue with any of what you say...

 

... the problem is that no digital synth is remotely built to audiophile standards (containing cheap DACs and components) and those using wave samples are usually samples that are compressed to death to save memory.

 

;-)

 

This is irrelevant because this is musical instrument - this is choice of artist, based on his preferences of sound tonality, not so much quality.

Recording need to be transparent after artist is created his art, or recording is part of creation (when someone wants some vinyl or tape specific sound). But then the final recording need to be maximally transparent.

 

Specific sound of instruments is part of creation, like sound of Moog synth or Hammond electronic orgel with all distortions they have.

Best recording brings this nuances to listener, without smooting anything.

Sorry, english is not my native language.

Fools and fanatics are always certain of themselves, but wiser people are full of doubts.

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As an owner of a large amount of synths I can't argue with any of what you say...

 

... the problem is that no digital synth is remotely built to audiophile standards (containing cheap DACs and components) and those using wave samples are usually samples that are compressed to death to save memory.

 

Synfreak's post about Plug-ins is also very revealing.

 

Just a a side note:

 

 

Many of the Plug-Ins which are used to "shape" the sound *) of a given PCM recording use internal oversampling.

 

 

So one always ends up with multiples of added (anti-aliasing) filters and lots of decimation processes ...

 

 

 

 

*) i.e.: compressors, limiters, EQs, ...

 

Now, does anybody believe that they all use the best oversampling filters?

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I think if I'm understanding correctly what people like tailspn are saying, the DSD/PCM conversion for editing can be done just to the portion of the file that must be edited. If that's the case, then presumably one wouldn't necessarily want to convert an entire file to PCM for listening because a few tenths of seconds had to be converted for editing.

 

Correct as stated when using Pyramix DAW's. But anyone mastering using a Sonoma or SADiE DAW does not convert to DXD (that's unique to Pyramix), but processes level changes in PDM at 64fs.

 

To Eloise's point, one must define what they mean when they talk about editing. Editing actually refers to splicing, the arranging of portions of takes on a timeline to produce a cohesive optimized music interpretation. It's one of many potential elements of post processing. Simple butt splices require no processing/conversations. Crossfades are actually continuous to some curve (usually cosine) of two overlapping takes. Crossfades in Pyramix requires DXD conversion for the time length of the cross. Sonoma and SADiE DAW's perform those in native PDM (8-bit Sony DSD-Wide).

 

What's more important to understand is recordings are made up of hundreds of spliced elements. Some of these elements may require effects processing, like level changes, mixing, EQ, reverb etc, and some may not. It's totally dependent upon the producer and artists, and their objectives for the project. So the recordings we purchase are all a hybrid of more or less process conversions through the time length of the recording. But those processes are typically applied where needed/wanted, and not the overall recording, and all at once one time when creating the edited master from all the accumulated contents on the Edit Decision List (EDL) and original takes.

 

That's of course using Pyramix. Sonoma and SADiE DAW's do the same, with far less available plug-ins (effects features), but all at the original 64fs bit-rate through remodulation.

 

The real point of all this is the question of what do customers want. And how many are in each of the potential market camps. Do the majority, or at least a sizable market share want as close as possible to the original analog mic feeds for their acoustic music, or want process sweetened candy sound, hyped up for perceived emotional satisfaction? The answer to that gets the processes, or lack of that I mention so briefly.

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Correct as stated when using Pyramix DAW's. But anyone mastering using a Sonoma or SADiE DAW does not convert to DXD (that's unique to Pyramix), but processes level changes in PDM at 64fs.

While I accept what you are saying ... PDM is not a format which can be distributed so requires some conversion.

 

Now If I understand correctly, what you feel is best is if there was some way to get PDM out of the ADC as currently you have to do is:

  1. take an analogue feed from a Microphone (or other analogue input)
  2. (inside the recording interface) this is converted to PDM
  3. (still inside the interface) this PDM has to be converted to DSD to output to the DAW
  4. (in the DAW) this is then converted to DXD or PDM for levelling and crossfades, etc.
  5. (still in the DAW) this is being converted back to DSD
  6. repeat 3-5 as required...
  7. a final conversion may be required prior to distribution

 

Now if there was some way to store PDM in a (standard) file, you could

  1. take an analogue feed from a Microphone (or other analogue input)
  2. (inside the recording interface) this is converted to PDM
  3. (in the DAW)levelling and crossfades, etc can be carried out in PDM domain
  4. convert for distribution in best quality to EITHER DSD or PCM

 

Now I'm not suggesting that the conversion stage looses much quality, but if it could be avoided then things would be even better?

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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That's a bit harsh. Anyone with half an ear will agree that CD (44.1 kHz) is far better than cassette tape.

 

A cassette master tape (mixed down from reel to reel tracks) was played back on a Nak CR-7a and used as a reference. The playback was captured at 88/24 and 44/16. I was not able to distinguish a difference in sound quality between playback of the cassette and the 88/24 digital copy. However, there was a clear degradation in sound quality with the 44/16.

 

Years ago, I did some tests with the same tape machine and with a 7.5 IPS 2 track machine using a live piano performance. With the cassette tape one had a choice: record with Dolby B or C and lose natural musicality, record without Dolby at a low level and capture the piano sound accurately, but with a high level of tape hiss, or record without Dolby at a high level and lose the dynamic impact of forte passages. With the open reel deck there wasn't a problem with finding a recording level that had acceptably low noise and still preserved dynamic impact. The reel to reel tapes were subsequently transferred to digital at 96/24 and the sound quality was distinctly superior to a digital transfer at 44/16. (When I began digitizing tapes I had assumed, based on theoretical arguments, that sampling at 44 kHz would suffice, given the limitations of the media. I was given the task of digitizing many recordings from a large archive and told to do so in 96/24. Since I was doing this work for free, I was going to push back on this apparent extravagance, but I decided to do some testing first, at which point it became obvious that 44/16 was not adequate for capturing the output of cassette tapes.)

 

I feel sorry for people who can not hear the differences involved, for one reason or other. It is their personal loss, unless they are advocating for continued use of mid-fi formats or, worse, if they are recording engineers who are degrading sound quality of the recordings they are producing. In the later two cases, I consider these people an enemy of good sound.

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If using a digital synthesiser to produce a recording, it ought to be possible to feed the digital signal directly into a DAW rather than going via an analogue path, in which case cheap DACs wouldn't be an issue. Maybe this is how it's done; I'm not involved in music production.

 

We'll you guessed it right in any case, that's what I do whenever the synth has digital out, it always sounds better.

 

;-)

Source:

*Aurender N100 (no internal disk : LAN optically isolated via FMC with *LPS) > DIY 5cm USB link (5v rail removed / ground lift switch - split for *LPS) > Intona Industrial (injected *LPS / internally shielded with copper tape) > DIY 5cm USB link (5v rail removed / ground lift switch) > W4S Recovery (*LPS) > DIY 2cm USB adaptor (5v rail removed / ground lift switch) > *Auralic VEGA (EXACT : balanced)

 

Control:

*Jeff Rowland CAPRI S2 (balanced)

 

Playback:

2 x Revel B15a subs (balanced) > ATC SCM 50 ASL (balanced - 80Hz HPF from subs)

 

Misc:

*Via Power Inspired AG1500 AC Regenerator

LPS: 3 x Swagman Lab Audiophile Signature Edition (W4S, Intona & FMC)

Storage: QNAP TS-253Pro 2x 3Tb, 8Gb RAM

Cables: DIY heavy gauge solid silver (balanced)

Mains: dedicated distribution board with 5 x 2 socket ring mains, all mains cables: Mark Grant Black Series DSP 2.5 Dual Screen

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Well, it is relevant in that it is the combination of both originating timbre and capture quality that produces the best recordings. If the artist cares about quality the chain begins with a head start.

 

The best recordings are those that closely mirror the artists intended outcome, the consumer then decides if they agree with the artists endeavours... but stuck in the middle of that perfect world there is a lot of very mediocre equipment, combined with numerous copying and then meddling by mastering engineers... and then there's the stupid A&R guy that wants his records to be louder than everyone else's.

 

It isn't that relevant to the core subject of this thread but it rather also mellows the argument, given how much damage is done before you decide what format to listen to.

 

This is irrelevant because this is musical instrument - this is choice of artist, based on his preferences of sound tonality, not so much quality.

Recording need to be transparent after artist is created his art, or recording is part of creation (when someone wants some vinyl or tape specific sound). But then the final recording need to be maximally transparent.

 

Specific sound of instruments is part of creation, like sound of Moog synth or Hammond electronic orgel with all distortions they have.

Best recording brings this nuances to listener, without smooting anything.

Source:

*Aurender N100 (no internal disk : LAN optically isolated via FMC with *LPS) > DIY 5cm USB link (5v rail removed / ground lift switch - split for *LPS) > Intona Industrial (injected *LPS / internally shielded with copper tape) > DIY 5cm USB link (5v rail removed / ground lift switch) > W4S Recovery (*LPS) > DIY 2cm USB adaptor (5v rail removed / ground lift switch) > *Auralic VEGA (EXACT : balanced)

 

Control:

*Jeff Rowland CAPRI S2 (balanced)

 

Playback:

2 x Revel B15a subs (balanced) > ATC SCM 50 ASL (balanced - 80Hz HPF from subs)

 

Misc:

*Via Power Inspired AG1500 AC Regenerator

LPS: 3 x Swagman Lab Audiophile Signature Edition (W4S, Intona & FMC)

Storage: QNAP TS-253Pro 2x 3Tb, 8Gb RAM

Cables: DIY heavy gauge solid silver (balanced)

Mains: dedicated distribution board with 5 x 2 socket ring mains, all mains cables: Mark Grant Black Series DSP 2.5 Dual Screen

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A cassette master tape (mixed down from reel to reel tracks) was played back on a Nak CR-7a and used as a reference. The playback was captured at 88/24 and 44/16. I was not able to distinguish a difference in sound quality between playback of the cassette and the 88/24 digital copy. However, there was a clear degradation in sound quality with the 44/16.

 

This only proves that 88/24 can transparently capture all the distortion and noise from the cassette, while 44/16 adds some audible artefacts of its own. It does not prove anything at all about the absolute fidelity of the cassette tape.

 

Years ago, I did some tests with the same tape machine and with a 7.5 IPS 2 track machine using a live piano performance. With the cassette tape one had a choice: record with Dolby B or C and lose natural musicality, record without Dolby at a low level and capture the piano sound accurately, but with a high level of tape hiss, or record without Dolby at a high level and lose the dynamic impact of forte passages. With the open reel deck there wasn't a problem with finding a recording level that had acceptably low noise and still preserved dynamic impact.

 

That's pretty much what I'd expect. Wider, faster tape performs better.

 

The reel to reel tapes were subsequently transferred to digital at 96/24 and the sound quality was distinctly superior to a digital transfer at 44/16.

 

I don't doubt this either. Looking at a 96/24 transfer from studio tape, it is clear that 44/16 would be inadequate. It is also clear that the 96/24 format still has room to spare. If you can show me a high-resolution tape transfer that exceeds 96/24 in either frequency or dynamic range, I'll accept the evidence. Of course, most of the available high-resolution files still fall short of that quality.

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