ASR Audio Science Review forum YouTube Channel

[The Computer Audiophile]

3 minutes ago, Samuel T Cogley said:

 

So you're in the "HQPlayer provides 'more accurate' playback" camp as well?  Doesn't it really depend on the source material and the DAC?  I've never seen HQPlayer marketed as "more accurate", and a quick check of the website confirms this.  Zero instances of "accura*"

 

Are HQPlayer's filters high quality?  No question in my mind.  But to make the leap of logic that this makes HQPlayer an "accuracy increaser" it a fool's errand IMHO.  It's precisely the flawed logic that MQA used.

 

EDIT: to the "accuracy", isn't this what the Closed Form filter is for in HQPlayer?  Do you like the way it sounds?

It’s all measurable. 
 

Think about it without HQP. Your DAC has filters. Without them the sound is horrible and inaccurate. 

[fas42]

32 minutes ago, Samuel T Cogley said:

 

 How could "more accurate" ever be measured?

 

The thing that really matters is "subjective accuracy" - this is having a lack of audible anomalies in the sound which irritate the brain; and which is currently essentially impossible to measure. You know when you have it when the recording goes the same, somewhat ho-hum, listening experience that you've had a hundred times - and becomes, 'magic' 😜 ...

[The Computer Audiophile]

42 minutes ago, Samuel T Cogley said:

 

How would you make the comparison?  What would you use as your "accurate" sample?  I think the whole "accuracy" trope is what got MQA in trouble in the first place.  I'll say again HQPlayer sounds better to me than any DSD DAC's builtin filters.  But more accurate?  How could "more accurate" ever be measured?

I say more accurate if it reduces things not on the original recording. Measurable things like aliasing in the audio band, distortions, increasing linearity, etc... All measurable. 
 

If things show up that aren’t on the recording and a filter removes them, I’d say it makes it more accurate to the recording. 

[Samuel T Cogley]

13 minutes ago, The Computer Audiophile said:

It’s all measurable. 
 

Think about it without HQP. Your DAC has filters. Without them the sound is horrible and inaccurate. 

 

I don't know Chris.  You guys seem pretty sure of yourselves.  I generally run my Yggy with unadulterated PCM.  When I use a DSD DAC, then HQP.  And that one limited use case (DSD to Yggy), I use HQP as well.  But no DSD DAC sounds as good as the Yggy to my ear.  And I'm able to sustain that elusive "EC" noise shaping now.  HQP is better than ever, but since there's no practical way to measure how "accurate" what I'm hearing is, it just seems like pomposity to declare my use of HQP makes my playback chain "more accurate".  Others might not hesitate though.  🙂 

[The Computer Audiophile]

3 minutes ago, Samuel T Cogley said:

 

I don't know Chris.  You guys seem pretty sure of yourselves.  I generally run my Yggy with unadulterated PCM.  When I use a DSD DAC, then HQP.  And that one limited use case (DSD to Yggy), I use HQP as well.  But no DSD DAC sounds as good as the Yggy to my ear.  And I'm able to sustain that elusive "EC" noise shaping now.  HQP is better than ever, but since there's no practical way to measure how "accurate" what I'm hearing is, it just seems like pomposity to declare my use of HQP makes my playback chain "more accurate".  Others might not hesitate though.  🙂 

No no no, somehow this got off on the path of all DACs all the time etc...

 

I started by saying some DACs. 
 

Your Yggy uses Mike’s super burrito DSP. Great stuff I think. It’s DSP and proves my original point about DSP equally more accurate sound. 
 

I don’t use any external DSP with my Yggy. 

[Samuel T Cogley]

3 minutes ago, The Computer Audiophile said:

No no no, somehow this got off on the path of all DACs all the time etc...

 

I started by saying some DACs. 
 

Your Yggy uses Mike’s super burrito DSP. Great stuff I think. It’s DSP and proves my original point about DSP equally more accurate sound. 
 

I don’t use any external DSP with my Yggy. 

 

To really put a point on it, Moffat makes accuracy claims that Miska does not.

 

 

[JoeWhip]

I am too dumb to be able to figure out HQP. Need a more powerful laptop to use it anyway, I use the newest Audirvana with my Yggy which I have set up to bit perfect and I let the Yggy do its thing. Sounds marvelous, even converting DSD to 24/176.4.

[The Computer Audiophile]

1 minute ago, Samuel T Cogley said:

 

Do really put a point on it, Moffat makes accuracy claims that Miska does not.

 

 

 

I think the whole point of HQP is accuracy for me. For others maybe not. 

[Jud]

42 minutes ago, The Computer Audiophile said:

I say more accurate if it reduces things not on the original recording. Measurable things like aliasing in the audio band, distortions, increasing linearity, etc... All measurable. 
 

If things show up that aren’t on the recording and a filter removes them, I’d say it makes it more accurate to the recording. 

 

I'd say measurement accuracy is what test signals are for in the first place, then whatever music you like to listen to when auditioning equipment to see whether your subjective self objects to anything your particular measurements may have missed.

[asdf1000]

8 minutes ago, Jud said:

 

I'd say measurement accuracy is what test signals are for in the first place, then whatever music you like to listen to when auditioning equipment to see whether your subjective self objects to anything your particular measurements may have missed.

 

This is how I use measurements. To shortlist what I demo and then decide with my own ears (subjective personal preferences).

 

I use various sources of measurements, ASR, JA at Stereophile, DS at SoundStage, HiFi News mag and more.

 

I mentioned this earlier in the thread, even before I asked for this thread to be closed on page 7 of this thread, because the thread had served it's purpose and was going in a direction I did not like ... that evidence is in writing below...

 

The Audiophile Stylist thought it was a good idea to keep the thread going, strangely.

 

 

 

 

[The Computer Audiophile]

Just now, asdf1000 said:

The Audiophile Stylist thought it was a good idea to keep the thread going, strangely.

Is that a personal attack?

[asdf1000]

8 minutes ago, The Computer Audiophile said:

Is that a personal attack?

 

How can it be when you've found it funny in the past?

 

[asdf1000]

4 minutes ago, The Computer Audiophile said:

Is that a personal attack?

 

How can it be when you've found it funny in the past?

 

[The Computer Audiophile]

8 minutes ago, asdf1000 said:

 

How can it be when you've found it funny in the past?

 

In many countries when a friend calls you a name it’s meaning is far different than when a foe calls you a name. 
 

How about in your country?

[lucretius]

57 minutes ago, The Computer Audiophile said:

 

I think the whole point of HQP is accuracy for me. For others maybe not. 

 

Iirc, HQP has a countless number of filters.  They cannot all be the accurate one?🙂  Secondly, how can a minimum phase filter, designed to change phase response, be described as "accurate"?

[Andyman]

In reverse order...

 

4 hours ago, The Computer Audiophile said:

P.S. @Andyman can you find someone on Earth who agrees with your logic on this? I'm all ears, but I think it's faulty. I'll even listen to the most objective people on this one. What does @pkane2001 say? I'm all ears. 

 

My logic is usually pretty sound (?!) However I fear you are conflating two subtly different arguments.

 

1) Whether playback via DSD upsampling is (inherently) superior (to plain pcm playback (upsampled to higher rate pcm or not)) 

 

Here the jury is still out but I raise you Mans (if he's still around - probably been banned for being a bit of a wanker at times but can be funny and knows a fair bit). Moffat too (two?) Probably mad Peter also but not entirely sure.

 

2) Whether playback via DSD upsampling provides a more accurate representation of what the artist (producer/engineer) intended

 

No it doesn't. This was my original argument (which I rather pompously labelled a philosophical point), so presumably where you (dare to) challenge my logic, and to which I still adhere (and is (surprisingly?) independent of 1) above)

 

It actually doesn't matter here, whether playback via DSD is or is not inherently superior (to plain pcm playback). It is sufficient that they are different.

Let me try it this way (rather than asdf100 or whatever he's called just repeating, copying and pasting over and over and over...)

 

For the sake of argument, let's agree that a playback chain which upsamples to DSD sounds different to one which does not.

 

Following months of toil and angst, an artist/producer/mastering engineer in the studio is preparing the final mix. Naturally there are no DSD dacs around here. The mix is perfected, being replayed via the ubiquitous pcm dac and signed off for release - phew, job done.

Now consider Chris C of this parish just happened by that day and inserted his upsampling to DSD gubbins (computer/dac - maybe even HQP 😳) into the mastering replay chain. We are agreed that the sound is now different (we could say we've applied a "DSD function")? However, if it was perfect before and is now different, it can no longer be perfect. It is by definition inferior - sadly disappointing (as mentioned earlier, it matters not one jot which replay chain is inherently "superior", merely that they differ. It may well be that the incredibly clever engineer could remaster the file so that replay via this latter DSD chain sounds identical to that of the former pcm chain. But that's of no relevance here).

The file is now released to the masses. Playback via my Gumby (upsampled to pcm384 or whatever probably similar to that in the mastering playback chain) sounds superlative.

Unfortunately Chris is still upsampling to DSD so what he hears has the aforementioned "DSD function" applied, thus detracting from perfection. (Of course had the cleverly remastered version been released, the roles would have been reversed - but they weren't). 

 

4 hours ago, The Computer Audiophile said:

 

 

Here's is where I believe you don't understand how digital audio works. 

 

All digital audio goes through DSP to get more accurate to the recording. Converting to DSD doesn't alter the sound away from the recording, it alters it closer to the recording. It doesn't matter what signal chain was used in the recording process, what matters is taking the delivered albums and playing it as perfectly as possible. 

 

For example, if you play PCM at 44.1 on a specific DAC, you'l get nonlinearities and distortions. These are added by the DAC and not in the recording. If you convert to DSD on that same DAC, you remove these nonlinearities and distortions through DSP and you can hear as close as possible what's on the delivered recording. 

 

By using DSP and filtering you get precisely what's on the recording. All DAcs work this way. It doesn't matter the process they use, PCM, DSD, R2R, Ring DAC, Sabre DAC, etc... it's about the final analog output. Using DSP and DSD on some DACs gives you more accurate analog output that enables you to reproduce the recording delivered to consumers, more accurately. 

 

 

 

How very condescending of you 😊. But it seems more likely you misunderstand. From above:-

 

Converting to DSD doesn't alter the sound away from the recording, it alters it closer to the recording

 

...so converting to DSD alters "it" (the sound?) closer to the recording. Presumably you understand the meaning of oxymoron?

 

And then:-

 

if you play PCM at 44.1 on a specific DAC, you'l get nonlinearities and distortions. [Maybe your talking here about NOS dac with no prior software upsampling? - hardly the norm]These are added by the DAC and not in the recording. If you convert to DSD on that same DAC, [errm - can't?!] you remove these nonlinearities and distortions through DSP and you can hear as close as possible what's on the delivered recording. 

 

C'mon chris admit it. You've had a chat with Miska, not really grasped what he's told you, and this is your unfortunate interpretation 😪!

 

Nevermind and no hard feelings.

 

Love,

 

Andy

 

[The Computer Audiophile]

Just now, lucretius said:

 

Iirc, HQP has a countless number of filters.  They cannot all be the accurate one?🙂  Secondly, how can a minimum phase filter, designed to change phase response, be described as "accurate"?

Are you suggesting better measurements and shifting noise way up into the inaudible band isn’t more accurate?

[Jud]

I see that our OP has taken a pretty bland pronouncement on my part about evaluating accuracy and used it as a tool to once again take the thread in the least productive direction possible.

 

I'd said I didn't intend to post further a while ago, but thought surely these friendly and reasonable exchanges could be made to serve no ill purpose.

 

Turns out I was wrong.  Perhaps it is the OP's intention to make their own thread of such low value that most people will leave and it will be effectively closed down, as they asked a while back.  With me at least, they have succeeded.

[The Computer Audiophile]

6 minutes ago, Andyman said:

In reverse order...

 

 

My logic is usually pretty sound (?!) However I fear you are conflating two subtly different arguments.

 

1) Whether playback via DSD upsampling is (inherently) superior (to plain pcm playback (upsampled to higher rate pcm or not)) 

 

Here the jury is still out but I raise you Mans (if he's still around - probably been banned for being a bit of a wanker at times but can be funny and knows a fair bit). Moffat too (two?) Probably mad Peter also but not entirely sure.

 

2) Whether playback via DSD upsampling provides a more accurate representation of what the artist (producer/engineer) intended

 

No it doesn't. This was my original argument (which I rather pompously labelled a philosophical point), so presumably where you (dare to) challenge my logic, and to which I still adhere (and is (surprisingly?) independent of 1) above)

 

It actually doesn't matter here, whether playback via DSD is or is not inherently superior (to plain pcm playback). It is sufficient that they are different.

Let me try it this way (rather than asdf100 or whatever he's called just repeating, copying and pasting over and over and over...)

 

For the sake of argument, let's agree that a playback chain which upsamples to DSD sounds different to one which does not.

 

Following months of toil and angst, an artist/producer/mastering engineer in the studio is preparing the final mix. Naturally there are no DSD dacs around here. The mix is perfected, being replayed via the ubiquitous pcm dac and signed off for release - phew, job done.

Now consider Chris C of this parish just happened by that day and inserted his upsampling to DSD gubbins (computer/dac - maybe even HQP 😳) into the mastering replay chain. We are agreed that the sound is now different (we could say we've applied a "DSD function")? However, if it was perfect before and is now different, it can no longer be perfect. It is by definition inferior - sadly disappointing (as mentioned earlier, it matters not one jot which replay chain is inherently "superior", merely that they differ. It may well be that the incredibly clever engineer could remaster the file so that replay via this latter DSD chain sounds identical to that of the former pcm chain. But that's of no relevance here).

The file is now released to the masses. Playback via my Gumby (upsampled to pcm384 or whatever probably similar to that in the mastering playback chain) sounds superlative.

Unfortunately Chris is still upsampling to DSD so what he hears has the aforementioned "DSD function" applied, thus detracting from perfection. (Of course had the cleverly remastered version been released, the roles would have been reversed - but they weren't). 

 

 

How very condescending of you 😊. But it seems more likely you misunderstand. From above:-

 

Converting to DSD doesn't alter the sound away from the recording, it alters it closer to the recording

 

...so converting to DSD alters "it" (the sound?) closer to the recording. Presumably you understand the meaning of oxymoron?

 

And then:-

 

if you play PCM at 44.1 on a specific DAC, you'l get nonlinearities and distortions. [Maybe your talking here about NOS dac with no prior software upsampling? - hardly the norm]These are added by the DAC and not in the recording. If you convert to DSD on that same DAC, [errm - can't?!] you remove these nonlinearities and distortions through DSP and you can hear as close as possible what's on the delivered recording. 

 

C'mon chris admit it. You've had a chat with Miska, not really grasped what he's told you, and this is your unfortunate interpretation 😪!

 

Nevermind and no hard feelings.

 

Love,

 

Andy

 

 

 

Andy, let's take HQP out of this and use only the DSP that goes on in every DAC. 

 

What's the purpose of this DSP? To remove things like noise, nonlinearities, and distortions among others. 

 

How is this not increasing accuracy? 

[Andyman]

11 minutes ago, The Computer Audiophile said:

 

 

Andy, let's take HQP out of this and use only the DSP that goes on in every DAC. 

 

What's the purpose of this DSP? To remove things like noise, nonlinearities, and distortions among others. 

 

How is this not increasing accuracy? 

 

Presumably the intention of all DSP is to improve something. But excepting those algorithms which preclude bit perfection, in what way do they differ with respect to accuracy? Mine's more perfect than yours?

 

No need to take HQP out of this. I have no doubt Miska knows what he's doing and why.

 

 

[lucretius]

36 minutes ago, The Computer Audiophile said:

Are you suggesting better measurements and shifting noise way up into the inaudible band isn’t more accurate?

 

No matter what format (PCM vs DSD) you digitize in, ultrasonic artifacts are created. The more bits you have, the lower the noise floor. Noise floor is lowered by roughly 6db for each bit.  DSD has significantly higher quantization noise than PCM, and the noise is much closer to audible frequencies, requiring significantly more sophisticated digital filters, as well as noise-shaping and upsampling algorithms.  When converting from PCM to DSD, more quantization noise and/or quantization errors are added to the recording.

 

Of course, using HQP to upsample DSD64 to DSD128 (and beyond)  is highly recommended because it puts DSD64/SACD quantization noise an octave above audible frequencies and allows better performing digital filters to be used. DSD128 has the majority of its quantization noise around 50KHz, which is fairly close to the same frequency as the majority of the quantization noise in a 44.1KHz PCM recording, which is centered around 44.1KHz.

 

[The Computer Audiophile]

1 minute ago, Andyman said:

 

Presumably the intention of all DSP is to improve something. But excepting those algorithms which preclude bit perfection, in what way do they differ with respect to accuracy? Mine's more perfect than yours?

 

No need to take HQP out of this. I have no doubt Miska knows what he's doing and why.

 

 

 

HI Andy, it will probably be easiest to digest this little pieces at a time, at least for me. 

 

I'm listening to a DAC that enables external DSP processing. When it's used, the DAC measures much better and is more linear. This provides a more accurate analog output through DSP. Removing nonlinearities is a good thing. It isn't a matter of mine is more perfect because the built-in filters can be bettered. No filter is perfect. 

 

 

 

 

[The Computer Audiophile]

4 minutes ago, lucretius said:

 

No matter what format (PCM vs DSD) you digitize in, ultrasonic artifacts are created. The more bits you have, the lower the noise floor. Noise floor is lowered by roughly 6db for each bit.  DSD has significantly higher quantization noise than PCM, and the noise is much closer to audible frequencies, requiring significantly more sophisticated digital filters, as well as noise-shaping and upsampling algorithms.  When converting from PCM to DSD, more quantization noise and/or quantization errors are added to the recording.

 

Of course, using HQP to upsample DSD64 to DSD128 (and beyond)  is highly recommended because it puts DSD64/SACD quantization noise an octave above audible frequencies and allows better performing digital filters to be used. DSD128 has the majority of its quantization noise around 50KHz, which is fairly close to the same frequency as the majority of the quantization noise in a 44.1KHz PCM recording, which is centered around 44.1KHz.

 

Agree!

 

I'm not advocating DSD all the time or PCM all the time or even more DSP all the time. I just believe that it can provide for a more accurate analog output in certain circumstances. 

[pkane2001]

17 minutes ago, Andyman said:

Presumably the intention of all DSP is to improve something. But excepting those algorithms which preclude bit perfection, in what way do they differ with respect to accuracy? Mine's more perfect than yours?

 

A reconstruction filter is a mathematical necessity for proper reproduction of frequencies below Nyquist/2. Filters can be leaky and even introduce phase and frequency artifacts. Many modern DACs reconstruction filters still pass through a lot of energy above 22.05kHz and this results in images in the audible band. I suspect most poorly filtered DACs are done so because it was cheaper and easier to implement. Some are "designer" filters meant to add something to the sound, rather than make it more accurate, and sometimes it's just incompetence. In these cases, upsampling or applying filters in software can make the DAC more accurate. 

[The Computer Audiophile]

1 minute ago, pkane2001 said:

 

A reconstruction filter is a mathematical necessity for proper reproduction of frequencies below Nyquist/2. Filters can be leaky and even introduce phase and frequency artifacts. Many modern DACs reconstruction filters still pass through a lot of energy above 22.05kHz and this results in images in the audible band. I suspect most poorly filtered DACs are done so because it was cheaper and easier to implement. Some are "designer" filters meant to add something to the sound, rather than make it more accurate, and sometimes it's just incompetence. In these cases, upsampling or applying filters in software can make the DAC more accurate. 

Exactly what my wife told me at breakfast this morning.