Help explain intersample overs, please?

[j_j]

What I'm getting at is that I'm sitting at my computer, where I listen via my Topping DX7 Pro, using its digital gain knob. It's never, ever close to the 0 dB setting, which would be painfully loud even it it didn't overdrive my iLoud MTM amps into saturation.

Unfortunately this also means that you're substantially raising the noise floor. If you're some place that that doesn't matter, fine. It would in my living room, and it would matter a lot in my studio.

It ALSO means that there will be problems with any kind of encoding (MP3, AAC, whatever) that isn't lossless, because that will push up the level even more, and since it may mildly modify phase, can create even worse overs, or outright clipping.

So, no, it's not harmless.

 

[earlevel]

Unfortunately this also means that you're substantially raising the noise floor. If you're some place that that doesn't matter, fine. It would in my living room, and it would matter a lot in my studio.

It ALSO means that there will be problems with any kind of encoding (MP3, AAC, whatever) that isn't lossless, because that will push up the level even more, and since it may mildly modify phase, can create even worse overs, or outright clipping.

So, no, it's not harmless.

I didn't say it's harmless. And I alluded to the noise floor. But here I am, 28 inches from my iLoud MTMs and I don't hear the noise floor (even if I get very close), despite being in a quiet studio on a quiet computer (Mac Pro 7,1—I paid dearly for that noise floor).

Yes, MP3, AAC encoding. But recall my words, the key to my post: "more and more", meaning as time goes on. Like I said, I listen to virtually all 24-bit. I'm not talking about the worst cases in life, I'm talking about the typical case, and what I deal with as an example. Again, I gave the caveat that I don't mean we shouldn't be concerned with the details. But as time goes on, I personally am confronted with fewer and fewer opportunities for inter-sample overs than in the past. I'm presenting the silver lining, not saying the cloud doesn't exist.

The only place that it could happen for me in a typical day is in my car, where the road noise alone obviates any discussion about things like digital noise floor. But since I don't have digital gain control there, that's the one place I could be assaulted by such overs. But, we both know that if I am, I'm unlikely to listen to anything where they are apparent. And certainly it's more of a problem that the speakers are in the door, and there is traffic and road noise.

 

[restorer-john]

@j_j Have you ever produced a series of files with graduated levels of intersample overs? A bit like the calibrated data interruptions on my test CDs.

I'd be very interested in testing a bunch of vintage players for the phenomenon and see what they are capable of resolving.

 

[j_j]

@j_j Have you ever produced a series of files with graduated levels of intersample overs? A bit like the calibrated data interruptions on my test CDs.

I'd be very interested in testing a bunch of vintage players for the phenomenon and see what they are capable of resolving.

I haven't but it's not hard at all. Just start with +1+1-1-1 and then reduce from 1 to maybe .2 in steps. Well, play them in the opposite order, see when/if it goes awry.

using four 1's and four -1's in a row also works, and can provide a more audible indication.

Using actual music material is a no-go because of copyright.

 

[restorer-john]

Using actual music material is a no-go because of copyright.

I give you full permission to use tracks from my debut album, copyright free - The Test Tones- Pure Sine Experience Volume 1.

 

[j_j]

I give you full permission to use tracks from my debut album, copyright free - The Test Tones- Pure Sine Experience Volume 1.

Well, you're welcome to my fugly "buzz tones" that are current lurking somewhere on Bob Katz's Site.

It's for finding IMD and nonlinear issues, and has found other "things" best left unheard as well. It doesn't do intersample overs, but it's discovered "things" some of which I am certain are wrong,but which I can not even begin to guess at the mechanism of. Suffice it to say 'linearity' is not the thing it finds, often.

 

[theREALdotnet]

Unfortunately this also means that you're substantially raising the noise floor. If you're some place that that doesn't matter, fine. It would in my living room, and it would matter a lot in my studio.

I don’t think it would matter to me. With noise+THD at -120dB these days I can afford to give away 20dB or so. I’d have to press my ear against the speaker to hear that noise.

A few weeks ago I played with very low-level tones (400Hz at -100dB generated by REW) to listen to the noise with different dither settings etc. I could only hear the tone right at the speaker, not from the listening position, and the noise, while audible, was extremely faint. The ear had to be in exactly the right spot at the drivers.

In short, a 20dB rise in the noise floor would likely never bother me. Headphone users may have a different perspective.

 

[restorer-john]

Well, you're welcome to my fugly "buzz tones" that are current lurking somewhere on Bob Katz's Site.

Had a hunt, but cannot locate anything. Only reference I found was this which didn't help:

 

[j_j]

Had a hunt, but cannot locate anything. Only reference I found was this which didn't help:

View attachment 257645

I'll go look.

 

[Hayabusa]

Probably obvious to a lot of people, but perhaps not everyone. More and more, modern convenience has made inter-sample overs simply a moot point.

What I'm getting at is that I'm sitting at my computer, where I listen via my Topping DX7 Pro, using its digital gain knob. It's never, ever close to the 0 dB setting, which would be painfully loud even it it didn't overdrive my iLoud MTM amps into saturation. And of course it's a similar situation with people using a laptop or other computer, with headphone or outboard speakers—you're probably using digital control. When I'm listening on my iPhone, the digital gain is rarely full blast except when I'm relying on hearing a conversation on the tiny built-in speakers—for audio listening through headphone and IEMS, it's backed off a few dB.

When working on music, I have a choice of always being at 0 dBFS and relying on analog pot somewhere, but it's more convenient to allow a bit of digital headroom and control it from sitting at the computer anyway. My 20 year-old Acura, which I fitted with a USB interface, is the only place where the phone feeds the USB with no ability to adjust the gain, and the only volume control is the analog one on the built-in stereo (yet it still manages to sound great—my music friends love to check out their mixes in my car, LOL).

But like I said, "more and more", it's unlikely for listeners to even generate such an over in normal listening.

NB: This is just an observation, I'm not saying that music production shouldn't strive to eliminate them, or that DAC manufacturers shouldn't pay attention. And of course I understand giving up a little dynamic range, but in general turning down is just dropping more into the noise floor anyway, and since virtually everything I listen to is 24-bit, there is no practical penalty for "throwing away bits" with digital gain control.

that should do it
But digital gain still does not guarantee your inter-sample overs are handled correctly but makes it much more likely....

 

[danadam]

I haven't but it's not hard at all. Just start with +1+1-1-1 and then reduce from 1 to maybe .2 in steps.

In attachment, 10x 0.5 second tones, the first one should have no overs, the following ones have 0.2, 0.4, 0.6, 0.8, 1.0 and then 1.5, 2.0, 2.5, 3.0 dB.

The spectrogram of it upsampled 2x:

 

[RichB]

I am no expert, but it seems to me that source material has been shown to have inters-ample overloads that many DACs have issues decoding.
Clearly DAC chip makers could handle this case but likely do not because it would reduce their S/N (SINAD).

Since ASR rates on SINAD, it is providing a perverse incentive to ignore this issue in favor of specsmanship.
Would it not make sense for ASR to measure an DACs handling of overloads to an incentive to improve or at least offer an option to handle this issue?

Overblown or not, overs are real and SINAD beyond a point is not.
Perhaps, SINAD playing a source with inter-sample overs, would make it applicable.

- Rich

 

[AnalogSteph]

Had a hunt, but cannot locate anything.

My guess is that they may be under Downloads, for which you need to have registered as a user first.

BTW, this relatively straightforward-looking website is about as responsive as the proverbial sloth dipped in molasses for me. I bet something is poorly optimized there.

Would it not make sense for ASR to measure an DACs handling of overloads to an incentive to improve or at least offer an option to handle this issue?

Definitely. While it has become less common to run DACs straight from optical disc players with no built-in volume control or volume normalization, it is still a valid use case.

A CD from peak loudness war times that I got recently showed peak levels of over 1.6 after a RG scan (with oversampling on). That's over +4 dBFS. Barely a track was substantially below 1.3 peak, still over +2 dBFS.

 

[bennetng]

I suppose here is JJ's "buzz" signal. Read the posts before and after it for more info.

Can You Trust Your Ears? By Tom Nousaine

CD players generally do a pretty good job of that. If you stick to mainstream recordings in mainstream formats, you are pretty safe. I've seen a few where the filter that is supposed to cut out the delta-sigma noise was pretty weak, and a couple DAC's that had 300+kHz tones leaking out. This...

www.audiosciencereview.com

...and here is a snippet of CDDA release with around +5dBTP. The whole album is in fact pretty quiet, but the contents are mostly chiptunes. Read the posts before it for more info.

Correlation between sample rate and audible frequency?

No -- it isn't ultrasonic. I don't understand why this misconception persists; perhaps some people are confusing noise shaping with ringing. Totally different concepts. The ringing is just an effect of taking out >fs/2 frequencies from the signal. They have no frequency components

www.audiosciencereview.com

 

[j_j]

I suppose here is JJ's "buzz" signal. Read the posts before and after it for more info.

Can You Trust Your Ears? By Tom Nousaine

CD players generally do a pretty good job of that. If you stick to mainstream recordings in mainstream formats, you are pretty safe. I've seen a few where the filter that is supposed to cut out the delta-sigma noise was pretty weak, and a couple DAC's that had 300+kHz tones leaking out. This...

www.audiosciencereview.com

...and here is a snippet of CDDA release with around +5dBTP. The whole album is in fact pretty quiet, but the contents are mostly chiptunes. Read the posts before it for more info.

Correlation between sample rate and audible frequency?

No -- it isn't ultrasonic. I don't understand why this misconception persists; perhaps some people are confusing noise shaping with ringing. Totally different concepts. The ringing is just an effect of taking out >fs/2 frequencies from the signal. They have no frequency components

www.audiosciencereview.com

That's a different one, but does the same thing. It's not for listening. Please!

 

[MusicNBeer]

I've tested 3 DACs for this, a Benchmark DAC1, Topping DX3Pro, and SMSL M500V1. All fail at full volume output. Luckily the latter two can be easily fixed by simply lowering the volume. The DAC1 has fixed level into the DAC chip so cannot be fixed unless you lower the input digital signal.

It's really dumb all DACs don't handle this as a routine use case of proper performance.

 

[theREALdotnet]

It's really dumb all DACs don't handle this as a routine use case of proper performance.

How would you like them to handle it, though? Apply a safe volume reduction on all input data? Turn on volume reduction only in case inter-sample overs are detected? Perhaps a warning LED indicating the condition should be commonplace. In the end, it is up to the user to maintain sensible headroom.

When done in software (on the players I’ve used anyway), delta-sigma modulation will be preceded by a safe volume reduction by default. In Audirvana, for example, it’s -3dB. You can set it anywhere between 0 and -6dB, but the default setting is there. If you know your signal has headroom (e.g. because you already provide it through volume levelling, as part of DSP, etc.) you can set this to 0dB. If you play straight rips from pop CDs you better don’t.

When delta-sigma modulation is done in hardware, how is the DAC supposed to know your situation?

 

[j_j]

How would you like them to handle it, though?

By just doing it right. Provide proper arithmetic and analog circuitry so there's no problem. It's not rocket science, it's not complicated, and by modern chip standards, it's not even expensive, in fact it's extremely easy to implement.

So just DO IT RIGHT!!!!

 

[MusicNBeer]

By just doing it right. Provide proper arithmetic and analog circuitry so there's no problem. It's not rocket science, it's not complicated, and by modern chip standards, it's not even expensive, in fact it's extremely easy to implement.

So just DO IT RIGHT!!!!

Exactly, this is part of a robust design that handles all reasonable inputs. Yes, the digital source engineers could do it, but they don't, so any competent DAC should account for it. The worst case is 44.1KHz so just give it 6 dB headroom. There's massive SINAD margin in modern DACs anyway. It's rediculous.

 

[restorer-john]

The worst case is 44.1KHz so just give it 6 dB headroom. There's massive SINAD margin in modern DACs anyway. It's rediculous.

When did you last see SINAD results for a modern DAC producing actual 16/44 content? Amir tests DACs with 24 bit data. Knocking 6dB off to protect against poor recording/mastering practices seems to me the wrong way to go about it.