Octave Music Don Grusin High Resolution Music Analysis (Video)

[Somafunk]

Steve Guttenburg has a 10 min interview with David Chesky regarding his new Audiophile society recordings including dsd and specific headphone mixes, be interesting to have Amir check them out.

David Chesky audiophile society here

 

[Miska]

Are these the "old" claims and if so how were they disproven exactly, apart from employing a complex high order modulator when a multi-bit modulator is an effective solution to maintain SNR?

Yes, exactly. And multi-bit modulator doesn't really make any notable difference. Those over 20 years old claims just demonstrate that if you create a poor modulator, the result is poor. You can also intentionally create a poor multi-bit modulator.

What you lack with multi-bit modulators at the moment is standard way to get those in and out of converters and standard delivery file formats. I don't mind myself, I'm all ready to utilize such immediately. Some 20 years ago I already implemented 8-bit modulator interfacing between software and hardware. But I'm not really a hardware manufacturer so that is only academically interesting.

Really, there's just no benefit to go through all the hassle.

 

[Miska]

In other word removing the bits information above 44.1kHz will change the audible waveform. You will not get what you want after the ADC. That hundreds megabytes wave are not to be touched, even you can't hear 80% of the information. If you touch it, it is 100% wrong.

It is not black and white. But DSD is efficient way to deliver non-bandlimited content.

 

[dc655321]

It is not black and white. But DSD is efficient way to deliver non-bandlimited content.

Given the bandwidth limitations of audio information receivers (hoomans), dsd has to be considered one of the most inefficient vehicles.

 

[Head_Unit]

ideally, the delivery should be multiple of 44.1kHz`:

Yeah it bugs me when this happens. Perhaps they are trying to hit the 96 and 192 buzzwords. Or on DVD itself, there was no 88/176, only DVD-A, but audio-only DVDs were ever only a tiny sliver so I doubt anyone is thinking about that any more.
- Then again, aren't there a number of DACs which also annoyingly don't support 88 or 176, even though they support 96 and 192?
- Or the goofy Apple TV which won't do 44.1...

 

[Head_Unit]

Why do they offer 24 bit audio when some of their DACs can't even break 16 bits of resolution anyway?

Shhhh, stop telling people the emperor has no clothes!

 

[kongwee]

Yeah it bugs me when this happens. Perhaps they are trying to hit the 96 and 192 buzzwords. Or on DVD itself, there was no 88/176, only DVD-A, but audio-only DVDs were ever only a tiny sliver so I doubt anyone is thinking about that any more.
- Then again, aren't there a number of DACs which also annoyingly don't support 88 or 176, even though they support 96 and 192?
- Or the goofy Apple TV which won't do 44.1...

DAC or ADC are in multiple of 48kHz anyway nowadays for decades. DAT is already 48kHz to begin with for studio.

 

[Head_Unit]

Slapping [DSD] on SACDs was a thing Sony came up with later.

Yes, and that was not really about sound at all. It was about continuing the revenue stream of CD whose patents I think were expired, and introducing something shiny and new to get more money and fight Napster et al, and COPY PROTECTION which was much more hardened on SACD.
There IS sonic truth to the idea of "well if the A/D is one bit then why not just store and play back the one bit?" - the thing is that is rather a lie since that rarely happened.
Huh, it is occurring to me that I don't recall seeing a DSD vs PCM comparison of A/D conversion, just playback. Anyone?

 

[whazzup]

In other word removing the bits information above 44.1kHz will change the audible waveform. You will not get what you want after the ADC. That hundreds megabytes wave are not to be touched, even you can't hear 80% of the information. If you touch it, it is 100% wrong.

Hmm.... you do know that to get a final complete track, the separate vocals / instruments / effects tracks have been converted / normalized / processed / truncated / mixed and 'corrupted' with god knows what plugins in the audio compositing software before reaching you right.... ?

With the above in mind, your statement about how the waveform is not to be touched is just...
Oh it has been touched all right....


Although it is possible you only listen to raw tracks and isolated vocals in your DAW.

 

[Michel Forbes]

Yeah it bugs me when this happens. Perhaps they are trying to hit the 96 and 192 buzzwords. Or on DVD itself, there was no 88/176, only DVD-A, but audio-only DVDs were ever only a tiny sliver so I doubt anyone is thinking about that any more.
- Then again, aren't there a number of DACs which also annoyingly don't support 88 or 176, even though they support 96 and 192?
- Or the goofy Apple TV which won't do 44.1...

As soon as you included video in the equation the sampling rate will be 48kHz.
Delaing with the .1 in 44.1kHz was a real pain in the ass during post-production.
It was decided with the venue of the Sony D1 transport (for video) to switch to 48kHz for PAL/NTSC video transport.
We have to keep in mind that in the 80,s the master for cutting CD was in fact a BVU 3/4 Inch U-Matic transport.
The PCM signal was recorded on video tape. (very interesting to watch...a full silence was black...any other modulation was black & white and moving).
DVD for video was based also on 48kHz.
The SACD was an incident...the anti-copy protection and the hassle to record/master and send the master to Sony before mastering for validation kill the medium.

But the dsd files are still available today.
A good DAC (properly clocked) will give an excellent playback of DSD files.

 

[Gringoaudio1]

Audio power amplifiers don't like amplifying ultrasonic crud; most will even protest with a constant 20kHz.

This is what I wondered. If your amp can amplify ultrasonic will it be heating up amplifying a constant noise signal at a relatively high level affecting the ability of the amplifier to perform well in the audio band?

 

[amirm]

Steve Guttenburg has a 10 min interview with David Chesky regarding his new Audiophile society recordings including dsd and specific headphone mixes, be interesting to have Amir check them out.

David Chesky audiophile society here

Cool. Will take a look. I have known David for two decades now and always appreciate how he experiments and pushes boundaries.

 

[Herbert]

DAC or ADC are in multiple of 48kHz anyway nowadays for decades. DAT is already 48kHz to begin with for studio.

24/96 established itself as a studio standard from 2000 on, based on 16/48 for broadcast/radio and film.

 

[dadregga]

Steve Guttenburg has a 10 min interview with David Chesky regarding his new Audiophile society recordings including dsd and specific headphone mixes, be interesting to have Amir check them out.

David Chesky audiophile society here

Worth a look, but https://theaudiophilesociety.com/pages/faq sounds like Second Verse Same As The First (emphasis mine):

We input 192/24, 96/24, DSD, and 48/24 into our mixers. No matter what the native rate is, we always mix and render the 3D sound as the highest possible sample rates to create these DSP environments. We have found in numerous listening tests a higher output sampling rate equates to a much larger and spacious soundstage, and creates more immersion, which is the goal of what we are doing. The object here is to use any device we can to increase the enjoyment of the final product for the listener. For these recordings, please listen with your ears, not a graph.

 

[Somafunk]

I hope not as the “listen with your ears” crowd fail to comprehend the many variables outside of our control when blindly listening with our ears, there is zero hope of a standard benchmark control.

 

[Miska]

But the dsd files are still available today.

And more coming all the time, mostly DSD256 recordings.

Huh, it is occurring to me that I don't recall seeing a DSD vs PCM comparison of A/D conversion, just playback. Anyone?

I'm using primarily two DSD ADCs, RME ADI-2 Pro and Merging Hapi.

As I said earlier, if you record 705.6/32 PCM or DSD256 with ADI-2 Pro, the results, including noise floor, are very similar.

Here you can see how ADC noise spectrum of ADI-2 Pro looks like when recorded using 705.6/32 PCM:

And here how it looks like when recorded using DSD256:

Almost identical. Which is not surprise given that the ADC chip is essentially two parallel DSD ADCs per channel.

 

[Herbert]

I hope not as the “listen with your ears” crowd fail to comprehend the many variables outside of our control when blindly listening with our ears, there is zero hope of a standard benchmark control.

The only variable is what equimpment they listened to while mixing and to what you are listening at home.
This variable is as old as the moment when Edison recorded "mary had a lamb".
The rest is just audiophile - well what bulls make. Also mixing for a headphone is nothing but a plugin that
mimics a listening room - NOT the stage it was recorded on.
For pros there are systems to mix multichannel on headphones: https://www.dear-reality.com/products/dearvr-monitor

 

[Head_Unit]

Almost identical. Which is not surprise given that the ADC chip is essentially two parallel DSD ADCs per channel.

Oh! I misspoke. Actually what you show is just what I was thinking...well maybe. If you start with one-bit at the A/D then the noise is going to stay the same even if you "PCM" it.
- Can you do the same comparison, but up to super high frequencies? (I'm kind of thinking the PCM will decimate off the highest frequency noise but my thinking on that is blurry today-I need to use MQA to "de-blur" my brain ha ha)

What I should have said is are there any non-hybrid pure multibit A/D converters any more? And if we measure those, how do they compare to 1-bit A/D?

 

[Blumlein 88]

Oh! I misspoke. Actually what you show is just what I was thinking...well maybe. If you start with one-bit at the A/D then the noise is going to stay the same even if you "PCM" it.
- Can you do the same comparison, but up to super high frequencies? (I'm kind of thinking the PCM will decimate off the highest frequency noise but my thinking on that is blurry today-I need to use MQA to "de-blur" my brain ha ha)

What I should have said is are there any non-hybrid pure multibit A/D converters any more? And if we measure those, how do they compare to 1-bit A/D?

I don't know if there were ever any of those for 24 bit application. There are some SAR ADC's though I'm not sure if any are used in audio anymore.

 

[Miska]

Oh! I misspoke. Actually what you show is just what I was thinking...well maybe. If you start with one-bit at the A/D then the noise is going to stay the same even if you "PCM" it.
- Can you do the same comparison, but up to super high frequencies? (I'm kind of thinking the PCM will decimate off the highest frequency noise but my thinking on that is blurry today-I need to use MQA to "de-blur" my brain ha ha)

Since highest PCM rate I can get out of ADI-2 Pro is 768k, it is not possible to compare to higher bandwidth. But I used matching rate families for this and not the 48k-base DSD although it works fine with ADI-2 Pro as well.

From noise perspective it doesn't make practical difference if it's like 1, 2, 4, 6-bit ADC or DAC. Modulators are blasting through full value range all the time. Sometimes multi-bit modulators are used in attempt to get away with simpler modulator implementation, but then you can usually see side-effects of it (can be observed especially with DAC chips).

If you convert either of above to 96/24 PCM, you see just a flat noise floor. If you convert to 192/24 you can see a little bit of lift in the noise floor above 50 kHz.

What I should have said is are there any non-hybrid pure multibit A/D converters any more? And if we measure those, how do they compare to 1-bit A/D?

Not on the market, have not been for many years. Last such was the said Pacific Microsonics Model Two. Which was used a lot for HDCD productions, technology from times before MQA, but technically similar. And it's maximum sampling rate was 176.4k. You can find some old recordings made with it from Reference Recordings for example, their newer ones are with DSD converters and such though. At least some of the 176.4/24 hires material you can find may be recorded with it.

Challenge for those are the similar you have for R2R ladder DACs. It is hard to get good precision and low level linearity. And you cannot use high oversampling rates because then you lose bits from the wordlength due to settling time issues, since converter needs to settle to +-½ LSB in fraction of sample period. This means you need more aggressive analog anti-alias filters which in turn means that especially your phase response gets bent.