Why 24/192 is a bad idea?

[Ethan Winer]

It's obvious to all of us Ethan that Barry has nothing to say to you so we all hope you get the message and drop it as Amir has so strongly warned you.

You're right, and I apologize for being argumentative. But I have to tell you that it's discouraging to post irrefutable proof about something, and not have that proof acknowledged. In this case I proved that the only thing affected by bit depth is the noise floor. So I'm satisfied my point was made, and I'll drop it. But it would be great if I didn't have to read the same myth again in this forum. If anyone doesn't understand the graphs I posted, and why they prove the point, please tell me what specifically is not clear and I'll try to explain further.

--Ethan

 

[treitz3]

You're right, and I apologize for being argumentative. But.....

You were fine up until the "but", Ethan. We are better than this here. Please hold yourself to a higher standard and open yourself to discussion and not confrontation or the provoking of hostility.

 

[Ron Party]

I want to hear whatever distortion was there when the talent approved it in the studio.

Interesting assumption, that is.

 

[rbbert]

You're right, and I apologize for being argumentative. But I have to tell you that it's discouraging to post irrefutable proof about something, and not have that proof acknowledged. In this case I proved that the only thing affected by bit depth is the noise floor. So I'm satisfied my point was made, and I'll drop it. But it would be great if I didn't have to read the same myth again in this forum. If anyone doesn't understand the graphs I posted, and why they prove the point, please tell me what specifically is not clear and I'll try to explain further.

--Ethan

Obviously there are plenty of people who don't think your "proof" in this area is "irrefutable"; you aren't going to convince them, and they aren't going to convince you. Interestingly, your post about that was on page 5 (out of 15, now) of this discussion.

 

[MarinJim]

Well, maybe you can help us by providing a short clip where 24/192 sounds audibly better than a 16/48 downscale from the same master?



Sorry to disappoint you but these days, most (all?) DACs I know about are actually sigma-delta ("1-bit") and they up-sample to the MHz range. Not that there's anything wrong with that.

I am with you 100%, but I think hi-res should be an option to all.

 

[MarinJim]

Well, maybe you can help us by providing a short clip where 24/192 sounds audibly better than a 16/48 downscale from the same master?

I di not say 16/48. I said 192 and 96



Sorry to disappoint you but these days, most (all?) DACs I know about are actually sigma-delta ("1-bit") and they up-sample to the MHz range. Not that there's anything wrong with that.

Danish Audio Design DAC 10 Tube Reference- NO upsampling, but will accept up to 192

"Ole Nielsen has chosen a classic and clean design which is based on the digital interface CS8416 Currus Logic which leads one DSD1792 per channel for converting the digital signal to analog. The digital part of the device can handle signals up to 24bit/192kHz but does not perform upsampling signals at a lower bitrate. The output current of each integrated d / a converted to a voltage by a circuit with operational amplifiers of the LM6172 National, who played the role of buffer / driver for both single ended and balanced for the exit."

 

[Ethan Winer]

Obviously there are plenty of people who don't think your "proof" in this area is "irrefutable"

So what part of my graph do you think is not conclusive? And why?

BTW, I am indeed trying to discuss.

--Ethan

 

[NorthStar]

I just finished reading this entire thread: fascinating exchanges of high technical merits,
with also great links provided. That's what a true audio forum should be all about;
discussions going forward.

* Personally I am a big Bob Stuart's fan (Meridian).
And high resolution audio with an "exchange rate/depth" of 20-Bit/88kHz, I believe, is sufficient;
and easier to implement, as I said in the past, and understood by the "understanders".

Last, I simply want to thanks Amir, Monty, and all the other participants,
as I am absorbing the great knowledge provided by few members here.

Bests,
Bob

 

[rbbert]

Danish Audio Design DAC 10 Tube Reference- NO upsampling, but will accept up to 192

Have you looked at the specs for the DAC chip itself? If it uses delta-sigma conversion (and maybe even if it doesn't), the DAC chip itself oversamples. Many manufacturers of entire DAC units use no oversampling filters prior to the DAC chip itself, and thus call their units NOS, but that's a bit misleading.

 

[jmvalin]

No I can't add it. You seem to be saying a lossy compression is lossless here. I want to hear whatever distortion was there when the talent approved it in the studio. Once you take out the ultrasonics that were there at the time the music was produced, I have no way of putting that signal dependent, originated from instruments, etc, back in. You can't unring the bell as the saying goes.

So it is not in this case about being better or worse but being as faithful as I can to what is the final product. Right now, if someone uses a tube pre-amp for the mic, there is distortion there that I also get to reproduce. In this instance though, I am told that I am better off hearing said mic without its tube distortion.

You're not getting it. This has nothing to do with faithful reproduction. Imaging a (rather simple) performance where at one point in time, all you have a 1 kHz tone, a 30 kHz tone, and a 33 kHz tone. In the studio, that's all you have, and that's what you record at 192 kHz. Now, when you play that back on your stereo, your speakers also get tones at 1 kHz, 30 kHz, and 33 kHz. Now, the problem is that your speakers are non-linear, so you also get IMD and the 30 kHz and 33 kHz now produce a 3 kHz tone. Compared to a 48 kHz (where only the 1 kHz tone remains), you can definitely hear a difference. But that difference is not the ultrasonic tones, it's the 3 kHz tone that was produced by the non-linear response in the speakers. That 3 kHz tone was never there in the studio. It's artificial.

Now, you could argue that it still sounds better -- just like a tube amp's distortion may sound better to you (that's an example, I don't want to discuss tube amps in further details). But in that case, you might as well add the distortion straight in the recording so that it sounds the same with any set of speakers.

 

[bdiament]

Hi MarinJim,

I for one, do not consider you wrong, and enjoy your posts. And I am glad you are advocating higher res. I must commend on your comment that if someone cannot hear hi-res, then by all means do not buy it. That said, I do not think h-res is a cure all to the digital world.An I do not tin it is "better" than the CD medium. I have heard recordings (I'm sorry I can't recall which ones) where to me 96 sounded better than 192 recording of the same material. What I am going to do (if anyone cares ) is buy a non-upsampling DAC that only does native processing, but to cover all my bases, can accept up to 192.

Thank you for your kind feedback.

When comparing the "same" recording in versions that are of different resolutions, it is important to be sure they are from the same mastering. In many comparisons between resolutions and formats, I've found the reality to be a comparison of two different masterings, at which point any comparisons of resolutions or formats is moot - we have what are essentially two different recordings, not two versions of the *same* thing.

This is part of the reason I created the Format Comparison page on the Soundkeeper site. Our releases are mastered once - at the same resolution at which they were recorded (24/96 for our first release and 24/192 for everything since). When the master is complete, we use iZotope's 64-bit SRC to create the 24/96 versions and to convert to 44.1k for the CD. Lastly, the CD gets its word length brought down to 16-bits and dither/noise shaping applied using iZotope's MBIT+ algorithm. (With a few dozen SRC and dither/noise shaping algorithms in the toolbox, these are the ones that I've found to consistently create results that sound the most like the originals.)

I've also found (and believe I've mentioned this elsewhere) that many converters with 192 in their specs actually perform better at lower rates like 96. I attribute this to the significantly increased demands on clocking accuracy made by the higher rates and by their increased demands on analog stage performance at wide bandwidth. Of the many converters I've heard, there have been few that, to my ears, reveal the true promise of 4x rates (i.e. 176.4k and 192k).

But with those converters that can truly do it, I find the results nothing short of magical. As I've said elsewhere, this is the first time in my experience the recording sounds to me like the mic feed. (The very best 24/96 I've heard, nice as this can be, is, to my ears, not close.) How this can be a "bad idea" will forever be beyond me. ;-}

Best regards,
Barry
www.soundkeeperrecordings.com
www.barrydiamentaudio.com

 

[MarinJim]

^^^ If you were me, that would be the best way to ensure 192 will be scrapped for 384 before the VISA bill was paid off...

The biggest variation in recording quality is IMO the source, mix, and mastering, not the resolution and rate of the playback. That said, I would hope hi-res recordings, like most higher-end products, reflect greater care in the recording and/or mastering process.

I have CDs of old records and it is clear that whomever did the CD mastering should be shot, or even worse forced to listen to the mix at realistic levels on a good system vs. a boombox or car stereo. Of course, they might not notice the difference...

Well, maybe you can help us by providing a short clip where 24/192 sounds audibly better than a 16/48 downscale from the same master?



Sorry to disappoint you but these days, most (all?) DACs I know about are actually sigma-delta ("1-bit") and they up-sample to the MHz range. Not that there's anything wrong with that.

To further promote your beliefs (and of many), what about even more? Say 386? Or higher?

 

[amirm]

You're not getting it. This has nothing to do with faithful reproduction. Imaging a (rather simple) performance where at one point in time, all you have a 1 kHz tone, a 30 kHz tone, and a 33 kHz tone. In the studio, that's all you have, and that's what you record at 192 kHz. Now, when you play that back on your stereo, your speakers also get tones at 1 kHz, 30 kHz, and 33 kHz. Now, the problem is that your speakers are non-linear, so you also get IMD and the 30 kHz and 33 kHz now produce a 3 kHz tone. Compared to a 48 kHz (where only the 1 kHz tone remains), you can definitely hear a difference. But that difference is not the ultrasonic tones, it's the 3 kHz tone that was produced by the non-linear response in the speakers. That 3 kHz tone was never there in the studio. It's artificial.

I fully got it and answered it. Once more, whatever happens in my home with 192 Khz, happened in the studio. If I am hearing a false 3 Khz, then they heard it too (or some facsimile of it). And based on that, approved a mix. Warts and all, I think people who pay a premium for music should get it all. Remember, if I got the original, I can always whack the ultrasonics should I find them harmful. But if they are truncated already, I am without options.

Now, you could argue that it still sounds better -- just like a tube amp's distortion may sound better to you (that's an example, I don't want to discuss tube amps in further details). But in that case, you might as well add the distortion straight in the recording so that it sounds the same with any set of speakers.

Even though you don't realize it, the last sentence is actually the situation at hand. The ultrasonics exist in the recording when it is originally mixed. It either is audible there or not. If it is audible, then I want the same music in my home. If it is not, then no harm is done. I have made this latter point so many times and have not seen you acknowledge it. Once again, what are you giving me for me to give up those bits?

 

[Bruce B]

I've also found (and believe I've mentioned this elsewhere) that many converters with 192 in their specs actually perform better at lower rates like 96.

We have found this as well. All converters have a "sweet spot" where they function better at one base sample rate (ie: 44.1 or 48). "Most" of the coverters I"ve used sound better at freq of 88.2/176.4/352.8

I attribute this to clocking within the unit.

 

[NorthStar]

We have found this as well. All converters have a "sweet spot" where they function better at one base sample rate (ie: 44.1 or 48). "Most" of the coverters I"ve used sound better at freq of 88.2/176.4/352.8

I attribute this to clocking within the unit.

-----Not multiple(s) of the standard 44.1kHz? ...And also.

 

[opus111]

But I have to tell you that it's discouraging to post irrefutable proof about something, and not have that proof acknowledged. In this case I proved that the only thing affected by bit depth is the noise floor.

Those two FFTs 'prove' no such thing.

If anyone doesn't understand the graphs I posted, and why they prove the point, please tell me what specifically is not clear and I'll try to explain further.

I do understand FFTs and how they can be misused. So do please go ahead and explain why they're 'proof' as you are claiming. Why would a couple of FFTs prove that 'the only thing that changed is the noise floor' ?

 

[RBFC]

It is not wise to make assumptions about the levels of intermodulation distortion created by playback chains downstream of the DAC. There are too many variables in play. If the original signal contains 30 kHz, etc tones, the we should get that on the recording and off the recording during playback. It makes no sense to me to eliminate a portion of the original signal due to concerns about its reproduction. New generations of technology always seem to address "huge" issues suffered by last year's gear.

Leave the recorded sound as it was played with minimal interference and let gear constantly improve to better extract the information accurately! Isn't this what has happened with redbook CD as digital technology began to mature?

If the original sound of the instrument produces intermodulation products because of its ultrasonic content, then that is the true sound of that instrument in nature.
Lee

 

[jmvalin]

I fully got it and answered it. Once more, whatever happens in my home with 192 Khz, happened in the studio. If I am hearing a false 3 Khz, then they heard it too (or some facsimile of it). And based on that, approved a mix. Warts and all, I think people who pay a premium for music should get it all. Remember, if I got the original, I can always whack the ultrasonics should I find them harmful. But if they are truncated already, I am without options.

I guess it depends on the definition of "in the studio". The 3 kHz tone is definitely not there during the recording. However, it may be there in the monitors when doing the mix. That being said, unless the monitors have exactly the same non-linearity as your speakers (highly unlikely), what you're hearing at home is not what they heard from the studio monitors.

Even though you don't realize it, the last sentence is actually the situation at hand. The ultrasonics exist in the recording when it is originally mixed. It either is audible there or not. If it is audible, then I want the same music in my home. If it is not, then no harm is done. I have made this latter point so many times and have not seen you acknowledge it. Once again, what are you giving me for me to give up those bits?

What makes you believe that your speakers have the same non-linearity as the studio monitors? After all, the power of the 3 kHz tone depends only on the speaker non-linearity. Interestingly, I haven't seen any speaker manufacturer include "ultrasonic non-linearity" in its specs. So maybe you're getting 10 dB more than what's in the studio monitors, or maybe 10 dB less. Who knows. OTOH, if you record the output of the studio monitors (or more simply, simulate the same non-linearity), then you can at least have hear the same thing.

 

[Orb]

^^^ If you were me, that would be the best way to ensure 192 will be scrapped for 384 before the VISA bill was paid off...

The biggest variation in recording quality is IMO the source, mix, and mastering, not the resolution and rate of the playback. That said, I would hope hi-res recordings, like most higher-end products, reflect greater care in the recording and/or mastering process.

I have CDs of old records and it is clear that whomever did the CD mastering should be shot, or even worse forced to listen to the mix at realistic levels on a good system vs. a boombox or car stereo. Of course, they might not notice the difference...

I agree that the biggest variation in recording quality is what you mention, but it is definitely worth mentioning every month Hifi News has at least one album that has issues with the resolution/rate of the playback when reviewing hi-rez.
Considering they only review about 5-8 hirez albums a month and are not looking for any that are known for issues this is a reasonably high issue.
Issues include both upsampling and downsampling (usually from SACD), and filtering related anomolies.
That said the very best 24bit hirez recordings do stand out as having great detail across the frequency range and amplitude, so again this ties into how well they are recorded-mastered for this specifically in the studio; seems many studios see this as a potential afterthought for hirez (when it should also be a primary consideration).
Of those 5-8 only a very few could be deemed as high quality recording mix-mastering for high rez (24/96 and higher).

I think this may be more obvious for them as they have developed their own analysis software that makes it much clearer than any of the standard software shown on here.
Downside is that these reviews-analysis are only available in the magazine publication and not on the Miller audio research site.
But they are great to look at and I find them very interesting when compared to the reviewers note on audio quality.

Cheers
Orb

 

[Orb]

We have found this as well. All converters have a "sweet spot" where they function better at one base sample rate (ie: 44.1 or 48). "Most" of the coverters I"ve used sound better at freq of 88.2/176.4/352.8

I attribute this to clocking within the unit.

Yeah totally agree and was one of my points earlier about implementation, even mentioned clocking (and potentially receiver chip used-implemented) as it seems only a few products these days are equal for both 44.1 and 48.
Also as I mentioned there are aspects of this that can be tied back specifically to the DAC chipset that have certain characteristics either or both for dbfs (one manufacturer has a pattern between -20dbfs and -40dbfs while another is around -90dbfs) or frequency range where the trend is after 20khz.
Cheers
Orb