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Comment by Derbasti | Hacker News Reader

He raises a lot of valid points. However...

192 kHz is clearly overkill for listening. Not so for further editing of the data.

Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible.

44100 is not a bad sampling rate, but it necessitates very sharp aliasing filters, which are audibly bad. A bit more headroom is well needed there.

That bit about intermodulation distortion is complete bogus. He talks about problems when resampling high-fs audio data. However, you would never do that. You would digitally process 192kHz all the way. Only your loudspeakers or ears would introduce a high-pass filter, and a rather bening (flat) one at that. There is certainly no aliasing going on there unless you resample (wrongly). Intermodulation distortion is not the fault of the sample rate.

I mayored in hearing technology. Calling 192/24 worse than 44.1/16 is total BS. How useful it is is a different debate.

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>Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible.

This [1] (widely accepted in the scientific audio community) study's conclusions disagree with your assertion.

>44100 is not a bad sampling rate, but it necessitates very sharp aliasing filters, which are audibly bad.

This is not the 1980s, hardware has progressed beyond that point. Modern (i.e. anything from 1995 onwards) DACs do not suffer from aliasing problems. Also see [1]

>That bit about intermodulation distortion is complete bogus. He talks about problems when resampling high-fs audio data.

I did not notice that in the article. It talks about IMD in the context of the analog chain and the transducers following the DAC, and it's possible that high frequencies can increase it.

[1] http://www.aes.org/e-lib/browse.cfm?elib=14195

> Modern (i.e. anything from 1995 onwards) DACs do not suffer from aliasing problems.

True, but they do so using (long, high-quality) high-cut filters. And these filters are pretty sharp, as they have to close within, say, 18-22.1 kHz. You can design them as linear-phase FIR filters with oversampling and all the good stuff, but physics dictates that sharp filters introduce distortion. A sharp filter like that is audible.

I'm not aware of any (blind) listening tests actually showing that a modern, high-quality DAC for 44 kHz audio introduces audible distortion compared to a similarly high-quality DAC for, say, 96 kHz audio, though. It's not theoretically impossible that the lowpass would introduce some sort of noticeable distortion, but I haven't run into substantiated evidence that it actually does.

44100 is not a bad sampling rate, but it necessitates very sharp aliasing filters,

When you're talking about recording, sure, but in terms of storage and playback, we solved that problem 20 years ago with oversampling.

You will still need a aliasing filter that cuts off between, say, 18 and 22.5 kHz to avoid aliasing noise. That is one sharp filter no matter how you look at it. You can use a high quality, long, linear-phase FIR filter, but you can't cheat physics: sharp filters necesserily introduce distortion, and such a sharp filter so close to the hearing threshold does not go unnoticed.

I don't see how a sharp filter could be needed if the DAC is oversampling.

Obviously. Analogue audio does not have a sampling rate. The ADC however can oversample all it wants, but if the output is 44.1 kHz, it needs an aliasing filter that cuts off at 22.05 kHz.

We appear to be in complete agreement with each other.

Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible

No, the difference is not audible at all. At 16 bits of depth on a normal low-level audio signal (~0.3 volts), we're talking about less than 0.000005 volts per amplitude step. This difference gets lost in the THD already at the DAC in your audio output stage. Then it gets lost again in the amplifier. And again in the cable to your speakers or headphones. And then it gets lost again in the speaker elements. What survives in a normal low-level audio signal is about 14 bits of resolution.

44100 is not a bad sampling rate, but it necessitates very sharp aliasing filters, which are audibly bad. A bit more headroom is well needed there.

44.1khz IS a bad sampling rate for accurately reproducing anything except a triangle wave or square wave above 5khz.

why do you think "This difference gets lost in the THD already at the DAC "? Do you have numbers to back it up? What's the noise floor of DAC? What's the noise floor of an output stage? Do you have the number?

The number is somewhere between the amplitude of an ant pissing on cotton, and an ant not even thinking about pissing on cotton.

high dynamic range is not about the lowest volume you can hear, it's about the voltage resolution between this sampling point and the next. Base on your assumption, we can all see black whether we use 16bit RGB color or 24bit RGB color, what's the point of using 24bit RGB?

Many years of building audio equipment (in particular analog synthesizers), and equally many years of being meticulously anal with getting the best components for my circuits, reading specifications of down to every single op-amp I've ever employed, is why I think so.

I am not saying that there aren't any DACs on the planet that can't handle five millionths of a volt, but I am saying that five millionths of a volt isn't surviving through the particular DACs and the rest of the electronics used in your PC/living room hi-fi audio equipment.

Heh, it's funny to see this late-nineties debate get re-hashed here. Also kind of fun.

If it were true that there's no audible difference between 16 and 24 bit, companies like Alesis, Otari, ProTools, etc. wouldn't have spent the last 15 years ditching 16 bit like an old pair of smelly sneakers. (better metaphors welcome).

Seriously, anyone who has sat down in a real listening environment for 5 minutes A/Bing 16 vs 20 bit, 16 vs 24, etc. hears the difference immediately. There's no question. This is why you can buy ADAT 16 bit 'blackfaces' for $100, down from their original $4,000.

Sure, moving up from 16bit recording was an improvement, but having done engineering for a company listed above for over a decade, I can tell you that we went 24bit/192kHz because of market demand, not for any real technical reasons. We thought it was fairly unnecessary ourselves. It was also kind of an arms race with other companies, much like the megapixel arms race for digital cameras.

Bigger numbers are better. Right?

It's all marketing, baby!

...And the new Pro Tools 10 just added the ability to record in 32-bit floating point. http://www.avid.com/US/products/Pro-Tools-Software

Yes, and anyone who has ever sat down infront of an LCD flatscreen watching their favorite movie on DVD/BD using gold-plated $200 HDMI cable instead of $4.99 Walmart HDMI cable see the extra sharpness immediately. This is why non-gold plated non-OFC HDMI cables are down to $4.99 a piece from their original $49.99 during introduction.

I can't tell if you're being sarcastic or not.

I’m going to go ahead and say yes, that seems to be blatant sarcasm, or at least, reference to placebo effect / being a sucker.

The difficulty I had is that the same person claimed they could hear the difference between 44 kHz and 96 kHz, when the article (and all other comments which cited outside sources) claims that is well outside of human capability.

That's cute. Obviously you've never recorded a rock band while riding the pre to compensate for 16bit's terrible noise floor and horribly limited headroom. You've never had the joy of ruining a perfectly good take because of that wonderful sound it makes when the volume spikes into digital distortion despite compressing the wazoo out of the input source. Glorious sound, digital distortion. Run a dentist drill through an old Speak & Spell and you'd just about have it.

You've never rented an expensive tube EQ during a mix to cover up 16bit's grating harshness from 10k to 15k. Or tried like mad to make the bass drum sound like a freaking bass drum and not a pie pan slamming against the back of a plastic trash can. And yes, we had good mics and pres, all standard studio stuff. Decent, not brilliant, converters, but it was the 16bit that was the problem. Getting those 20bit XTs for the first time was like walking into the Promised Land.

Sure, there's lots of marketing ploys out there, lots of snake oil. Moving up from 16 bit was not one of them.

Did you read the original article at all?

> Professionals use 24 bit samples in recording and production [11] for headroom, noise floor, and convenience reasons.

...snip...

> Modern work flows may involve literally thousands of effects and operations. The quantization noise and noise floor of a 16 bit sample may be undetectable during playback, but multiplying that noise by a few thousand times eventually becomes noticeable. 24 bits keeps the accumulated noise at a very low level. Once the music is ready to distribute, there's no reason to keep more than 16 bits.

The original article does say that yes, during recording and production, 24 bit audio gives you a lot more room to play with. That doesn't mean that you can hear the difference between 16 and 24 bits for the final recording; just that 24 bits give you more room to keep out of trouble during production.

Did you read the comment thread at all? I wasn't responding to the article, I was responding to a comment:

...snip...

>Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible

No, the difference is not audible at all.

...snip...

It looks like you are jumping in without actually having read the article in question. That's ok, but you are wasting space building a straw man proceeding to vigorously demolish him.

The original article explicitly mentions how 24bit is useful for recording.

Speaking of jumping in without reading...I wasn't responding to the article. I was responding to the commenter that said you couldn't tell the difference between 16 and 24 bit.

And you cannot tell the difference. The reason to record using 24 bits is so you don't have to be as precise centering the recording level. If that level is centered then you can capture fine with 16 bits (by the way that is also explained in the article).

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