Room Correction Hardware/Software Issues, Options. Please Advise.

[dped90]

Maybe something for Merging to consider.

Or not, despite mentioning that the NADAC was twice above my budget.....

RE: [MT General Inquiry] Hapi Tutorials for Audiophiles, Long Overdue

Info Europe Merging <[email protected]>

Hi Greg,
Thanks for the mail. As you correctly mentions, the Hapi is not destinated at audiophiles, though we have a lot of user e.g. on the Hapi user forum. As it is not for audiophies and we are not trying to target this market, there are no intention of making specific videos for audiophiles. The market for a non adapted product is too small and we had the NADAC range in the past focusing on this market and users (now being replaced).

All the best

 

[Keith_W]

"Now being replaced" is interesting. I wonder what they are planning.

 

[dualazmak]

@dualazmak is running a system like this. I have tagged in so that he can weigh in.

Thank you for your kind attention on my multichannel audio project.

In the end, all of us place different weightings on acceptable sound, cost, convenience, learning curve, and so on. The problem with forums like ASR and posts like mine is that we all speak with our own biases on how WE place the weightings which is the source of a lot of disagreements (for example, I don't think differences between DAC's is that important). You need to weigh up your priorities first

For me, too! I agree with you...

Hello OP @dpeg90,

Have you already decided to go forward in the direction of "Merging Technology HAPI 8-CH or 16-CH multichannel DAC with Ravenna routing"? If this would be the case, I need to say nothing for it since that direction would be the best way at present for your needs, even though rather expensive; and we know @Kal Rubinson has now fully implemented it in his audio rig, and also (hopefully!) he will soon kindly give us his detailed review on it.

Only in the case if you have not yet fully decided the direction with 8-CH DAC unit, as @Kal also suggested in his above post #3, I would like to suggest you to purchase/try OKTO DAC8PRO in rather reasonable/acceptable budget; if you still would like to go into "Merging Technology HAPI", then you may easily release your DAC8PRO to the used market (even within ASR) having acceptable (still nice) price tag, I believe.

Furthermore, (I know this is my own bias, as @Keith_W indicated, though), I highly recommend you to try/test an independent DSP software having excellent GUI within your PC or Mac ("EKIO" in my case on Windows PC) as system-wide one-stop DSP Center which can receive all the audio signals from audio-output software like audio(-visual) players (JRiver MC, Roon, Audacity, Adobe Audition, etc.) and also from web browsers (Chrome, Edge, Firefox, etc.).

The use of excellent ASIO/VASIO/VAIO routing software "VB-Audio Matrix" will enable all the multichannel I/O for your test and validation; the very nice feature of gain adjustment on each of the routing grids in Matrix routing grid screen fully resolves the known major issue of gain matching between your local music library play with JRiver and Youtube listening with web browser(s). Under "VB-Audio Matrix" routing, you can select its virtual VAIO input channels as Windows default audio playback device in upto 192 kHz processing (ref. here and here on my project thread).

If you would be interested in my PC-DSP-based multichannel multi-SP-driver multi-amplifier fully active audio project, using VB-Audio Matrix, EKIO and DAC8PRO, you can find my latest system setup here and here on the project thread. Please note that I (we) can process the DSP (EKIO) and DAC (DAC8PRO) in upto 192 kHz which is more than enough at least in my audio project (ref. here).

 

[dped90]

Thank you for your kind attention on my multichannel audio project.



For me, too! I agree with you...

Hello OP @dpeg90,

Have you already decided to go forward in the direction of "Merging Technology HAPI 8-CH or 16-CH multichannel DAC with Ravenna routing"? If this would be the case, I need to say nothing for it since that direction would be the best way at present for your needs, even though rather expensive; and we know @Kal Rubinson has now fully implemented it in his audio rig, and also (hopefully!) he will soon kindly give us his detailed review on it.

Only in the case if you have not yet fully decided the direction with 8-CH DAC unit, as @Kal also suggested in his above post #3, I would like to suggest you to purchase/try OKTO DAC8PRO in rather reasonable/acceptable budget; if you still would like to go into "Merging Technology HAPI", then you may easily release your DAC8PRO to the used market (even within ASR) having acceptable (still nice) price tag, I believe.

Furthermore, (I know this is my own bias, as @Keith_W indicated, though), I highly recommend you to try/test an independent DSP software having excellent GUI within your PC or Mac ("EKIO" in my case on Windows PC) as system-wide one-stop DSP Center which can receive all the audio signals from audio-output software like audio(-visual) players (JRiver MC, Roon, Audacity, Adobe Audition, etc.) and also from web browsers (Chrome, Edge, Firefox, etc.).

.......If you would be interested in my PC-DSP-based multichannel multi-SP-driver multi-amplifier fully active audio project, using VB-Audio Matrix, EKIO and DAC8PRO, you can find my latest system setup here and here on the project thread. Please note that I (we) can process the DSP (EKIO) and DAC (DAC8PRO) in upto 192 kHz which is more than enough at least in my audio project (ref. here).

Thanks for sharing these options. One question comes quickly to mind: As Kal Rubinson confirmed that the Hapi DA8P card does not do DSD upsampling of lower resolution sources like CDs, do you use any software that does?

The Okto DAC 8 Pro is an excellent option and with my main speakers being > 94db/meter efficient there should be no chance of gain loss regardless the choice of RC software. But I wish I could be as sure of this with my Rythmik F12 subwoofers. That's why the Merging Hapi looks like the safer bet; given its 6.9 v unbalanced/13.8v balanced maximum outputs, and at presumably low distortion levels-however I do hope that Stereophile's John Atkinson will confirm this when he does his measurements on the Hapi during Kal Rubinson's review of same.

What I'm also hoping for is that since Merging turned down my request to produce video tutorials to simplify setting up an 8 channel Hapi for 2.x or 5.3 surround use in home, I wish some ASW members would do so; walking dummies like me step by step through the procedures-thereby avoiding all of that
irrelevant Hapi bloatware.

 

[dualazmak]

Thanks for sharing these options. One question comes quickly to mind: As Kal Rubinson confirmed that the Hapi DA8P card does not do DSD upsampling of lower resolution sources like CDs, do you use any software that does?

Sorry, but I do not fully understand what would be your point and concern with "DSD upsampling".

Would you please carefully read my post here #532 on my project thread, and please get back to me if needed;
- Summary of rationales for "on-the-fly (real-time)" conversion of all music tracks (including 1 bit DSD tracks) into 88.2 kHz or 96 kHz PCM format for DSP (XO/EQ) processing: #532

The Okto DAC 8 Pro is an excellent option and with my main speakers being > 94db/meter efficient there should be no chance of gain loss regardless the choice of RC software. But I wish I could be as sure of this with my Rythmik F12 subwoofers.

I also do not understand what would be your point and concern on actual gain/volume of room air sound at your listening position. I believe this would the role of your preamplifier and/or power-amplifier (not the task of DAC unit) for which you can select suitable amplifier(s) based on the policy of "right-person-in-right-place" to drive each of the SP drivers (ref. here).

As you may know well, in multichannel multi-amplifier setup, we do not need powerful amplifier for tweeters and supertweeters, but rather small-power low-distortion high-S/N amp should be selected. (e.g. ref. here).


Furthermore, let me remind you that "where to control master volume/gain" would be really critically important point in multichannel multi-amplifier setup, as you may know. In my setup,

Even though I can control "master volume" at most upstream JRiver MC's "internal volume" or "DSP EKIO's input channel gain" or "DAC8PRO's internal digital master gain/volume", I always prefer (and I believe it is the best) using the most upstream JRiver MC's "internal volume" as system's safe and reliable master volume controller. Please read and understand carefully my latest system setup shared here on my project thread, especially this total signal path diagram;

Just for your convenience, you can find here and here the "Hyperlink Index" for my project thread.

 

[dped90]

Actually, my knowledge on this is limited and fragmented. But Kal Rubinson and most are aware of what how DACs which can upsample to DSD 256 can often make CD tracks sound better. The first person to inform me of this is Markw4. See his 6/17/23 reply to me here.

https://www.diyaudio.com/community/conversations/benchmark-dac-3.986505/#convMessage-1106636

 

[dualazmak]

But Kal Rubinson and most are aware of what how DACs which can upsample to DSD 256 can often make CD tracks sound better.

Really??
I feel you are insisting as if a low resolution photo (600 x 600 pixel) could be upsampled into 5120 x 5120 pixel photo with better resolution and image quality.

Even though I myself definitely do not agree on this, if you would like to do so, even JRiver MC (not a DAC) can upsample/encode on-the-fly (real-time, while listening to music) any of the PCM music track into upto 8xDSD in native format to feed it into 8xDSD capable DAC. In this case, further DSP processing (XO/EQ/Group-Delay/Relative-gain) on such DSD signal in digital domain in native DSD format would be impossible.

 

[dped90]

Really??
I feel you are insisting as if a low resolution photo (600 x 600 pixel) could be upsampled into 5120 x 5120 pixel photo with better resolution and image quality.

Even though I myself definitely do not agree on this, if you would like to do so, even JRiver MC (not a DAC) can upsample/encode on-the-fly (real-time, while listening to music) any of the PCM music track into upto 8xDSD in native format to feed it into 8xDSD capable DAC. In this case, further DSP processing (XO/EQ/Group-Delay/Relative-gain) on such DSD signal in digital domain in native DSD format would be impossible.

First, it was Markw4 (and probably many others) but not me who claimed that at least in some cases upsampling a CD track and feeding it to a DAC with DSD256 capability will yield better sound quality. Indeed, any such improvement is likely just a psychoacoustic effect, as upsampling cannot add bit depth nor increase the sampling frequency of any 16 bit/44.1kHz recording.

However, am I correct that your speakers are actively crossed and done so by your room/system EQ software? If yes, if I did attempt to use JRiver, Foobar or another Windows player to do such upsampling and fed to the Okto DAC 8 Pro-which does have DSD256-since my speakers are passively crossed would I have any problems using REW or Audiolense for room/system EQ?

 

[dualazmak]

However, am I correct that your speakers are actively crossed and done so by your room/system EQ software? If yes, if I did attempt to use JRiver, Foobar or another Windows player to do such upsampling and fed to the Okto DAC 8 Pro-which does have DSD256-since my speakers are passively crossed would I have any problems using REW or Audiolense for room/system EQ?

Yes, my system-wide software DSP Center "EKIO" can handle upto 192 kHz PCM for its DSP (XO/EQ/Group-Delay/Relative-Gain) processing for unlimited numbers of I/O chanel. My OKTO DAC8PRO can handle upto 192 kHz PCM for 8-CH in-sync simultaneous DAC processing using its ESS 9028PRO DAC chip which has digital total/individual gain controllers.

I usually upsample (or downsample), therefore, all of my digital library music tracks (majority of them are 44.1 kHz 16 bit CD ripped tracks) into 88.2 kHz or 96 kHz for DSP processing by EKIO; I can do it upto 192 kHz, if I would like to do so (again ref. here #532).
- Summary of rationales for "on-the-fly (real-time)" conversion of all music tracks (including 1 bit DSD tracks) into 88.2 kHz or 96 kHz PCM format for DSP (XO/EQ) processing: #532

In case if you would try/test signal chain of JRiver MC --> EKIO --> DAC8PRO --> multiple-amplifier --> multiple-SP-driver, you may quite easily establish all-in-ASIO routing within Windows 11 PC by using the wonderful and robust VB-Audio Matrix (a reasonable donation software, USD 5.0 or more, completely free for first month); ref. here #858 on my project thread.

BTW, you may of course implement upstream DSP in PC, even if you would still like to use single stereo DAC and single 3-way (or 4-way) passive (LCR-network) SP system as shown in this diagram;

 

[Keith_W]

Thanks for sharing these options. One question comes quickly to mind: As Kal Rubinson confirmed that the Hapi DA8P card does not do DSD upsampling of lower resolution sources like CDs, do you use any software that does?

The Okto DAC 8 Pro is an excellent option and with my main speakers being > 94db/meter efficient there should be no chance of gain loss regardless the choice of RC software. But I wish I could be as sure of this with my Rythmik F12 subwoofers. That's why the Merging Hapi looks like the safer bet; given its 6.9 v unbalanced/13.8v balanced maximum outputs, and at presumably low distortion levels-however I do hope that Stereophile's John Atkinson will confirm this when he does his measurements on the Hapi during Kal Rubinson's review of same.

What I'm also hoping for is that since Merging turned down my request to produce video tutorials to simplify setting up an 8 channel Hapi for 2.x or 5.3 surround use in home, I wish some ASW members would do so; walking dummies like me step by step through the procedures-thereby avoiding all of that
irrelevant Hapi bloatware.

I personally think that DSD upsampling will not give you much audible benefit, but there are plenty of people who do. It is possible to use software to convert PCM to DSD and upsample it in real time, the only software I am aware of that can do this is HQPlayer. If you do use HQPlayer, you need to be aware of the following caveats:

- HQPlayer is expensive! USD$326.
- Performing 6-8 channels of conversion + convolution is EXTREMELY CPU intensive. It is not uncommon for people to use dedicated PC's for HQPlayer, and these are typically fire breathing watercooled monsters with high end graphics cards (HQP takes advantage of nVidia CUDA cores) typical of gaming setups.

If you decide to use HQP, you can use either the embedded version (where it lives on its own dedicated PC) or the player version (where you use a third party playback program like Roon on the same computer). Again, there are people who believe that separating the convolution/upsampling function from playback into two PC's results in better sound. I am not one of them.

Don't forget that the thermal cost of using HQP is significant and you will need a strategy to deal with that. If you decide to air cool your PC, your CPU fans will be running all the time and it will raise your room's noise floor unless you move your PC to another room and run a long network cable to send output from the PC into your Merging via Ravenna. Or you could consider watercooling, which does reduce noise, but increases the complexity of your build.

Now, I know for a fact that my Merging NADAC has a lower SINAD when fed a DSD signal (-118dB) compared to 44.1kHz PCM (-110dB). I have measured it with an E1DA. However, upsampling PCM from 44.1kHz to 192kHz also brings the SINAD down to about -116dB. The CPU cost for upsampling PCM 44.1kHz --> PCM 192kHz is minimal, my CPU typically sits at 8% usage when I am doing this. OTOH, upsampling PCM 44.1kHz --> DSD128 results in CPU usage of 95%, and I sometimes get audible signal dropout. The benefit is a 2dB improvement in SINAD. I can tell you that I can not hear a difference between PCM 44.1kHz and PCM 192kHz, and I definitely can not hear a difference with DSD. The reason I upsample to 192kHz is because it is a "free upgrade" in SINAD and it only costs me a tiny bit extra CPU usage and raises my electricity bill by a tiny amount.

I spoke earlier in this thread about our own biases, and this is one of mine. IN MY OPINION, DSD upsampling is not needed. If it came with a low CPU cost, I would do it and grab the extra -2dB improvement in SINAD even though I know perfectly well it is not audible. What you are really doing is expanding a lot of effort into pushing an inaudible number into an even more inaudible number. There is plenty of low hanging fruit to pick before you even do that. I can tell you for a fact that creating better convolution filters through improvement in my understanding of room acoustics, measurement methods, knowing what to correct/what not to correct, and better usage of my software (Acourate) results in massive objective and subjective improvements to the sound.

However, there are people who want "the ultimate" and build two PC's, upsample to DSD, electrically isolate all network connections by using optical, use atomic clocks, audiophile grade switches, and the like ... here is an example of such a system. I applaud his effort, determination, and attention to detail. In fact I have to say that I admire him, even if I disagree with half the things he is doing. You can decide if you want to go down that path, but I doubt that many people on ASR would think that going to such lengths is fruitful.

 

[Sokel]

Correct me if I am wrong, but all ESS DAC's convert PCM to DSD internally.

I don't think they do.
One of them at least does things very differently between PCM and DSD:

PCM vs DSD

Did a little comparison and I was wondering if any friend with better gear here has tested this. Results you're about to see is probably narrowed to the specific DAC but it's interesting to see how it handles things differently. I will not try to decipher the results other that DSD seems much...

www.audiosciencereview.com

 

[dualazmak]

Hello again OP @dped90,

I do hope much good luck in your DSP audio exploration whatever direction you would select to go!

Now, let me suggest two points before your actual first step-forward in your exploration.

1. You would please establish your reference/basic non-DSP audio setup with an amplifier and your passive SP system(s) to which you will be easily rollback anytime for objective and subjective comparative measurement/listening during your step-by-step DSP multichannel journey. In my case, OPPO stereo SONICA DAC +Accuphase E-460 integrated amp + passive Yamaha NS-1000 + Fostex T925A supertweeter + L&R Yamaha YST-SW1000 subwoofer have been always my "reference" audio system throughout my long (about 5-year) multichannel exploration.

2. Although I do not know your music preference/genre, I highly recommend you to prepare your own consistent "Audio Reference/Sampler Music Playlist" consists of at least 30 music tracks of excellent recording quality and hopefully having some kind of objectively proved measurement data on sound quality; whenever you would proceed one-step forward with your DSP-based system, you need to vary carefully subjectively listen (and objectively measure) the improvement (or not) using your consistent "Reference/Sampler Music Playlist". Just for example and for your possible interest, you can find summary of my such "Audio Reference/Sampler Music Playlist" here #669 and here #670 on my project thread.

 

[dped90]

I personally think that DSD upsampling will not give you much audible benefit, but there are plenty of people who do. It is possible to use software to convert PCM to DSD and upsample it in real time, the only software I am aware of that can do this is HQPlayer. If you do use HQPlayer, you need to be aware of the following caveats:

Thanks for your very thorough reply and for advising me against what would clearly be foolish behavior. Again, PCM to DSD conversion for upsampling is something I was told which could, in some cases, give better sonic results than doing PCM only upsampling. Prior to that I thought that any DSD features a DAC has are usable only for playing SACDs or track rips from same. In any case, while the Xeon CPU in my pc packs a lot of horsepower no way would I resort to any kind of upsampling that would generate high temps and cause unacceptable fan noise-which could likely occur due to the convolved filter being applied to several DAC channels. And if HQplayer would also exploit my Nvidia card’s cores power draw, heat and fan noise levels would rise further.

Furthermore, even if I had access to free or nearly free solar electricity, I happen to be an a/v enthusiast with an environmental conscience. Minimizing heat and noise emissions, along with my electricity bill, are nearly as essential to me as implementing system/room EQ. Thus, the miniscule benefit and consequences of pursuing PCM to DSD conversion, according to your observations, would be wholly absurd.

But as you claim that you could also hear no improvement when doing PCM upsampling of 44.1kHz content to 192kHz then I fail to see the logic in doing that either.

 

[Keith_W]

But as you claim that you could also hear no improvement when doing PCM upsampling of 44.1kHz content to 192kHz then I fail to see the logic in doing that either.

As mentioned, the reason I upsample from 44.1kHz to 192kHz is because it comes with minimal increase in CPU usage, and I get a "free" -6dB improvement in SINAD. There is absolutely no good reason for me to do it apart from feeding the worst of my audiophile OCD tendencies. I can't hear a difference, but the knowledge it's there makes me feel warm and fuzzy.

In fact there are a lot of things in my system that are overkill and which I don't need. Acourate has 131,000 taps. I don't need that many taps. I can time align different drivers down to a difference of a few millimeters. I don't need to do that, but I do it anyway. And my 96dB/W/m horns don't need 100W of amplification, but since I already have the amplifiers, that's what I use. The list goes on.

I think there is a difference between actively pursuing something that you know will make a marginal difference, and already having the capability of an "on paper" improvement. It costs me next to nothing to upsample PCM 44.1kHz to 192kHz. But it will cost me a lot of money and inconvenience to build a watercooled PC to run HQPlayer.

 

[dualazmak]

I think there is a difference between actively pursuing something that you know will make a marginal difference, and already having the capability of an "on paper" improvement. It costs me next to nothing to upsample PCM 44.1kHz to 192kHz.

Yes, essentially agree with you; within my PC too, the on-the-fly upsampling of PCM 44.1 kHz to 88.2 kHz, 96 kHz, 176.4 kHz or 192 kHz by JRiver MC has very little additional workload to CPU.

By the way, I once read (but I do not remember the exact source) that, for upsampling of PCM 44.1 kHz (majority of my digital music library), the integer multiples, i.e. upsample to 88.2 kHz/176.4 kHz, would be better/recommended; what would be your thoughts on this?

I know some of the DAC chip have separate internal hardware/software for DAC process of 44.1/88.2/176.4 series and 48/96/192 series, but I know little about the internal processing by software DSP in this regard; maybe, I assume essentially no difference (e.g. 88.2 kHz vs. 96 kHz, 176.4 kHz vs. 192 kHz) in 64 bit floating point IIR filter calculations (ref. here and here) and other DSP tasks within DSP "EKIO" in my setup; of course I have no audible difference at all, though.

 

[dualazmak]

I can time align different drivers down to a difference of a few millimeters. I don't need to do that, but I do it anyway.

Yes, I did the same!

Very fortunately in my case in my listening environment, the 1.0 msec precision time alignment between L&R subwoofer and woofer (actually 16.0 msec) plus 0.1 msec precision between woofer and others (midrange+tweeter+supertweeter) (actually 0.3 msec) well contribute to audibly better total sound quality/3D-perspectives (ref. here and here).

I would like to suggest OP @dped90, therefore, trying time alignment between all the SP drivers at the final stage of his DSP implementation even though intensive objective measurements and subjective listening would be needed.

 

[Keith_W]

By the way, I once read (but I do not remember the exact source) that, for upsampling of PCM 44.1 kHz (majority of my digital music library), the integer multiples, i.e. upsample to 88.2 kHz/176.2 kHz, would be better/recommended; what would be your thoughts on this?

I don't have an opinion on it because I don't know. We need someone with more knowledge of how digital works, like @solderdude. Or maybe you could start a separate thread?

 

[dualazmak]

I don't have an opinion on it because I don't know. We need someone with more knowledge of how digital works, like @solderdude. Or maybe you could start a separate thread?

Thank you, understood well. I assume it would not be the major issue at least for me worthwhile to start a new thread on this topic...

In any way, during my daily music listening sessions, I usually upsample/downsample all the tracks to 88.2 kHz, and it always sounds really nice as shared here on my project thread.

 

[dped90]

Thank you, understood well. I assume it would not be the major issue at least for me worthwhile to start a new thread on this topic...

In any way, during my daily music listening sessions, I usually upsample/downsample all the tracks to 88.2 kHz, and it always sounds really nice as shared here on my project thread.

Again, I want to express my thanks to Keith W and dualazmak for laying bare the false advantages of PCM to DSD conversion, while enumerating every horrific effects it would have on the system I'm poised to build. The other good news is that most of dualazmak's sources are CDs, and who apparently gets wonderful results by PCM upsampling CD tracl rips to 88.2kHz. Btw, do you rip to uncompressed WAV or FLAC. Which file format, and why? If you save to FLAC files do you run check sums to look for bit errors in the rip?

BUT here's my current concern along with a special test request: Many here and at other forums are aware of intersample overs distortion-the primary cause of it and how deliberate (loudness wars) or accidental (pre-loudness wars) it's been often found on music CDs, and perhaps also in movie and TV show audio on DVDs and BDs. Benchmark Media's John Siau probably gives the most concise explanation.

Intersample Overs in CD Recordings

Intersample Overs are a Common Occurrence in CD Recordings We have frequently used Steely Dan's Gaslighting Abbie from Two Against Nature in our listening tests. This is a spectacular CD recording with lots of dynamics and a low noise floor. Nevertheless, in a little over 5 minutes, this track...

benchmarkmedia.com

But the subject probably gets the most thorough examination here.

https://gearspace.com/board/mastering-forum/1401406-intersample-clipping-audible.html

Whatever side of the issue you may find yourself on, since it will be months before I would get close to making this test myself, if you happen the Steely Dan album Siau referenced
https://en.wikipedia.org/wiki/Two_Against_Nature (or your local public library may have a copy), please
rip the track "Gaslighting Abbie".

The track runs 5:53. Siau said that very shortly after 5 minutes the distortion begins cropping up. Listen carefully from ~ 4:50 onwards and note how the sound quality will likely change. But wWhether or not you hear any distortion, upsample it to 88.2kHz. During that playback was any distortion you heard before upsampling now gone or reduced to any degree?

 

[Keith_W]

Again, I want to express my thanks to Keith W and dualazmak for laying bare the false advantages of PCM to DSD conversion, while enumerating every horrific effects it would have on the system I'm poised to build.

I think I was careful to note that there are very small measurable improvements for choosing to go DSD or upsampling depending on what DAC you are using.

The other good news is that most of dualazmak's sources are CDs, and who apparently gets wonderful results by PCM upsampling CD tracl rips to 88.2kHz. Btw, do you rip to uncompressed WAV or FLAC. Which file format, and why? If you save to FLAC files do you run check sums to look for bit errors in the rip?

It was such a long time ago now that I have forgotten why I chose FLAC! I think the major reason for me is that FLAC allows you to include album art and more metadata. Some people claim to hear differences between WAV and FLAC. I haven't bothered to do that experiment because (1) I doubt if it's true and (2) even if it's true I don't think I want to know, because it took me years to rip my 3-4000 CD's into FLAC! Lol!

I used Exact Audio Copy to rip all my CD's to FLAC.

Whatever side of the issue you may find yourself on, since it will be months before I would get close to making this test myself, if you happen the Steely Dan album Siau referenced
https://en.wikipedia.org/wiki/Two_Against_Nature (or your local public library may have a copy), please
rip the track "Gaslighting Abbie".

Sorry, I don't have that disc.