Dynamic range question, Dynamic range of hearing vs. equipment Options

[thesurfingalien]

Hi All,

I am having problems understanding the concept of dynamic range, and more specific how the dynamic range of human hearing relates to the dynamic range as specified for audio equipment.

What I have read on the Net is that the dynamic range of human hearing is, on average, 135 dB (the pain limit / threshold?).

I also understand that the dynamic range of a CD (16 bit) is limited to 96 dB, and for HD (24 bit) recordings 144 dB. This dynamic range is calculated by the bit-depth.

For playback of music (I use 44.1/16 flac only) I use a E-MU 1820 that has a dynamic range of about 115 dB (or so). I am not sure what my amp can do...

However, when I play music I hardly pass 80 dB (or so) listening volume, so the 96 dB CD data (if the recording makes full use of the range that is) gets "compressed", while the relative dynamic range remains.

After the long intro my question in essence is:

Does the human hearing have a similar "resolution" that is comparable to the resolution defined by the bit-depth in (digital) audio? Or am I just comparing apples and pears?

Regards,
Peter

[dc2bluelight]

Very interesting discussion. I'd like to add a couple of things.

The 96dB FS to Noise ratio of a 16 bit system is theoretical. It's almost never exactly 96dB, though. Dithering shaves off 3dB or so, then the so called "noise shaping" D/A converters put 8dB or so back in, if that's even real. But FS is a maximum peak figure, we measure noise using an RMS meter with (hopefully) a weighting filter in front of it, either A or ITU-R468, either of which will tend to push the reading downward making 96dB look more like 103 or so.

Rooms, on the other hand, generally have their noise figures expressed as NC (Noise Criterion) figures. NC is a set of curves, and to qualify for a given NC curve, each octave band of measured noise must fall below that standardized curve. So, the actual measurement of room noise is done flat, but the result has to fit a curve that allows lots more energy at the low end. It's not an A weighting curve, so it's not directly related to a SNR figure of a digital system. And, though I can't site a reference, it's been shown recently that the "average" living room hovers around NC-25. By the way, you can't actually measure below an NC-20 with a 1/4" measurement mic. Too much self noise., you need a 1" mic with a quiet pre to do it.

Also, when we say we listen to our music at 80dB SPL, again, that's an average level, and might be measured with an average meter (probably) or an RMS meter. Depending on how a CD is mastered, the average levels might be as much as 15dB - 20dB below FS, or as little as 8dB below FS (hope not, but they crunch 'em pretty hard these days). It's not unreasonable to expect to listen at 80dB SPL (average, unweighted) and still get a peak or two at 95 - 100dB SPL. Depending on your SPL meter ballistics, you may or may not see that peak or read it properly, though.

It would still seem that 16 bits is more than enough to handle a 95dB SPL peak in an NC-20 room.

There is an advantage to recording at 24 bits. For one, if you're recording live, you've got (theoretically!) 48dB more room to make a level error in. If you're mixing, or processing, you've got more resolution in which to adjust gain, process, eq, etc. But there's a really big rub: Almost without exception, 24 bit sound cards NEVER produce 144dB of dynamic range! Just check the specs of almost any "affordable" sound device,oh heck, even the unaffordable ones. In fact, I've found exactly one A/D that is commercially available that has a real 24 bit's worth of dynamic range: http://www.stagetec.com They get 153dB A, and don't even bother with a mic preamp! Everything else is WAY less.

Which is the point. Why would to stretch your system to handle 24 bits when even the so-called 24 bit material has a dynamic range of more like 18 - 20 bits?

So I'm saying 16 bits is enough for "release" material, perhaps a little shy for capture. And I'm reminded that ratio of 0dB SPL to the Threshold of Pain is about 22+ bits.

[Arnold B. Kruege...]

In fact, I've found exactly one A/D that is commercially available that has a real 24 bit's worth of dynamic range: http://www.stagetec.com They get 153dB A, and don't even bother with a mic preamp! Everything else is WAY less.

The means that they use to obtain the 153 dB dynamic range has been around for a long time.

They basically run a bunch of converters in parallel. Each converter has an input attenuator with a different amount of attenuation. They then pick an output from a converter that is neither clipping nor close to the bottom of its range, and scale it digitally based on that converter's input attenuation.

For example, let say we have two medium-priced converters with 110 dB dynamic range. We connect one to the source with no attenuation, and put a 30 dB attenuator in line with the other converter. We then monitor the status of the two converters. In general we use the output of the unattenuated converter, but when it gets within 3 dB of clipping, we switch over the the converter with the 30 db attuator. On the digital side we multiply the digital output of the attenuated converter by 30 dB in the digital domain. When we switch over to the output of the unattenuated converter we simply use its digital output as is.

The net result is a system with about 140 dB dynamic range made out of out of two $5 ADC chips, and a few other relaitvely inexpensive parts. We do have to pay attention to the recovery time of the unattenuated converter, because it may take a few milliseconds to recover from clipping when the input signal gets too high and it clips.

The other approach to extending the dynamic range of conveters is to simply run them in parallel and obtain a 3 dB noise reduction every time we double the number of converters. I've never seen more than 8 convertors used this way.

Neither of these approaches reduce nonlinear distortion. They just address noise.