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Digital Audio and Amplifier Noise Floor Comparison - Is 16bit/44.1kHz All We Need ???


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24 minutes ago, Paul R said:

 

You are welcome to pick anything you want to measure by.  All I said was that 24bit is below the noise floor of most equipment, and 16bit is above same. Exactly how much is dependent upon equipment chosen, but in general, that holds true. 

 

I am unclear exactly what your objection is. Here are a couple other amps. 

 

The now $379 Outlaw Audio M2200 amp is 112db A weighted, and still falls within that criteria. 

The $1700  Peachtree Nova 150 I use a lot is 105db A weighted. 

 

I just thought it was interesting that you used spec from the Benchmark amplifier which was about where you said ordinary equipment should measure.

 

As for your own amplifier, the A weighting is likely to be giving a big improvement in the spec since it's a class D amplifier.  Not to be down on your amp, just to point out what is realistic.  If you don't care about the A-weighting of the amp spec, then you shouldn't care about noise shaping with 16 bit to get much better SNR.

 

Edit: In other words, I suggest that if you are using A-weighted spec to try to match equipment to 16-bit audio, then you should use the effective SNR for dithered 16-bit, so something like 115 to 120 dB.

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19 minutes ago, Paul R said:

 

A weighted specs are often all that is easily available, but that is a good point. What would you suggest are figures you would be happier with? I generally prefer A weighted specs myself.

 

However,  it does not follow that noise shaping to boost the 16bit noise floor is the same thing, or even related. I think you are conflating two different subjects there.

 

 

I agree that my numbers might not make sense.  I meant more to make the general point.

 

I don't worry about it so much myself.  If I could afford the benchmark I would buy it, just because it is a great amp not because I think I need 130dB.  And if I owned that amp I wouldn't then go out and seek 24 bit audio; I'd still be happy with 16-bit.

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4 hours ago, esldude said:

A-wtd is usually a 4-6 db improvement over no weighting.  So it isn't like unweighted results are suddenly terrible.  More common is fudging on real values or some trickery in gain staging that gives a number you won't see in normal use.  As Rt66indierock said, your noise floor is much closer to the signal at 1 watt levels.  Of course with common speakers you only need mid-80 db SNR to be so low it will never be heard at 1 watt.  Plus your SNR with dithered 16 bit is worse by a half bit to bit.  So rather pointless to engage in petty arguments without putting all this in context. 

 

I've found it interesting no one ever points out that with shaped dither and 16 bit audio there is a real benefit to high sample rates.  Here is where I took silence, and saved it in 16 bit with shaped dither in both 44/16 and 192/16.  I then filtered out everything above 20 khz in the 192 file.  Notice the shaped 44 has a noise level of -84 db while the filtered 192 is -120 db.  This is what would happen if your speakers (and your ears) don't respond above 20 khz.  Of course no one does 192 at 16 bit do they.  Getting back on topic, something like this could have been done with MQA without requiring licensing, messing with undecoded fidelity or causing other problems.  

 

image.thumb.png.c044ec0125a9c1be6a0ab0ac2892cd97.png

 

This is interesting, but if the goal is increasing SNR then it is a pretty inefficient way of doing it, compared to just going with greater bit depth.

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18 hours ago, sandyk said:

Hi Richard

The Benchmark is not alone in giving decent Unweighted and Weighted noise figures these days , with even many kit amplifiers performing very well at a cost, typically including metalwork , transformers etc. of <Au $1,000 

 Even the recent less ambitious Silicon Chip magazine designed 200W SC200 Amplifier specifies Signal-to-Noise Ratio: -116dB unweighted with respect to 135W into 8Ω(20Hz-20kHz)

I have attached the specs for the previous more ambitious SC ULD3 amplifier (123dB Unweighted)

from March 2012, as well as their DAC from way back in Sept.2009 which is capable of further improvement using more recent voltage regulators etc.

 Undoubtedly there are numerous commercially available amplifiers and DACs that easily surpass the performance of these older designs these days.

 

Regards

Alex

SC ULD 3 p.1- Specifications.jpg

SC DAC PART 1 -p.3a.jpg

Just out of curiosity, have you or @Shadders seen measurements of finished amplifiers made from these kinds of kits?  I'm not sure I've seen measurements of any amplifier other than the AHB2 that meets the 96dB THD+N at 1W.  Maybe nCore is right on the edge of meeting this with a 22KHz filter.  Maybe there are a few others, but I bet they are rare.

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1 hour ago, Shadders said:

Hi,

Sandyk can confirm this, but the specification for the SC200 was measured for the completed amplifier - not simulation. The ULD was also the measured performance. If you purchase the kits/PCB, and follow the article instructions, then you will achieve the measured performance.

Regards,

Shadders.

I can't find any numbers or curves showing THD+N at 1W for that amp.  The neuorochrome modulus stuff also looks like it may meet the 96dB at 1W, but the unit reviewed on ASR is not quite there.

 

Edit:  The ASR test of the neurochrome amp  is only slightly worse than the manufacturer curve (apparently due to intentional enclosure compromise), and this is into 4 ohms.  THD+N into 8 ohms is better for this amp - I think it may meet the 96dB at 1W mark.

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25 minutes ago, esldude said:

You do realize with most speakers being about 86 db for one watt you only need -86 db to get everything down to 0 db SPL when measuring at 1 watt.  When taking into account Fletcher-Munson you can probably have complete perceptual silence with even less.  I'm all for good performance, but the idea you need THD+N this low is a little over the top for 1 watt.  And for most good amps you really are going to being talking about N(oise).  

It's a great point.  Likewise, this is the kind of thinking that makes sense regarding whether dithered 16 bit audio gives enough resolution - consider realistic peak SPL and room noise.

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Thanks for making the new thread.  I vote Yes as well.  Maybe amplifiers can be better than 16 bit equivalent, but doesn't room noise kill the possibility of realizing that much dynamic range in practice?

Edit: I didn't notice the 44.1.  That one is a little more tricky I'd say, but my vote would still be yes.

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11 minutes ago, esldude said:

Thinking a sound level meter shows noise levels in my listening room of say 35 db SPL means that sets a lower limit is deceptive.  Most room noise is heavily weighted toward low frequencies.  If you split the bands up like our ears do, in our most sensitive frequencies 3-5 khz most rooms are anywhere from 0 db SPL to 10 db SPL in that range.  Some treated studios can be -10 db SPL in that range. We also can hear into noise 10 or 15 db as well.  So as a basic starting point I would suggest we assume 0 db SPL as our in room noise floor. 

That's a really good point.  But do really think 0 dB is a realistic assumption.  What is the perceived level and FR of normal breathing, for example?

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4 minutes ago, sandyk said:

 

 Why ? Many people listen to music at quite high SPLs, often in excess of the requirements of the source material.

Why limit this to just RBCD when many people even listen to TV via their main system, and that may also include music videos .

 With TV audio the max. level is limited anyway to suit the medium, but dialogue etc,. is often listened to at higher than the original voice levels. 

 

I'm not sure I am following you.  Are you saying it is nice to have extra amplifier dynamic range to account for level mismatches when paired with multiple sources?  I can buy that argument.

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4 minutes ago, Hugo9000 said:

Isn't there a meme about hearing people breathe?  Like "if I can hear you breathing, I have fantasized about your death" or something?  Or is it chewing that bothers most people?  I personally don't want to hear chewing or breathing!

 

Sorry for the OT!

But if you hear your own breathing, what does that mean?

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Keeping it simple - if playing a silent track from your highest level source (or maybe your preferred source) at full volume is not audible, doesn't that mean your amp has all the SNR that you need?

 

Doesn't address the 16 bit question, though.  Also, nothing wrong with wanting an amp that has much more than enough SNR.

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8 hours ago, sandyk said:

I disagree.

 I have been able to hear a small but worthwhile improvement from the listening position after further reducing the already very low noise level contribution from a DC Offset Corrector, which wasn't audible from the listening position, and really only JUST visible at the Preamp's Output at maximum gain and a 1K resistor across the Input, with  a CRO at maximum gain PRECEDED by a Low Noise 10 x gain battery powered Preamplifier !

Even though you were trying to reduce noise, and did so, how do you know that the improvement that you are hearing can be attributed to noise reduction, and not an improvement in something else like some form of distortion?

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On 6/21/2019 at 7:20 PM, sandyk said:

psug

I can only go by what I saw on the screen of the C.R.O both before and after the changes. The original DC offset corrector from 1987  that mine was based around  used a TL071 I.C. and you could also see and hear the improvements simply by changing to lower noise opamps.

Initially I changed to a AD744, then finally a low noise OPA134.  Another constructor of my design suggested that I change from a 2n2 capacitor across R8  to a  4n7 as he had noted a further small improvement;. I also noted a further minor improvement, although the C.R.O. was unable to confirm this visually.

Any residual noise from the Offset Corrector goes into the base of a transistor in the differential  input pair.

Incidentally, this amplifier and the Preamp versions have a distortion of <.0006 %, so I doubt that you would readily  hear any further reduction in distortion figures.

 I use DC Offset Correctors because both my Class A 15W/Ch. and Class A Preamp/HA are fully DC coupled.

i.e. there are no input capacitors or a capacitor in the gain setting areas.

Alex

new offset--VRegs added.jpg

Tillbrook Offset Corrector.jpg

Did you only change the op amps in one of the circuits, or did you replace the bottom circuit with the top circuit?  I would think it is possible that what you hear is related to better power supply rejection, or from the top circuit providing more complete removal of the AC in the input.

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13 hours ago, sandyk said:

The bottom circuit was part of the original design from 1987 that I constructed. There weren't that many low noise opamps to choose from back then. My own Class A fully DC coupled preamps and 15W Class A were constructed many years later ,where I realised that the original circuit could be markedly improved in order to reduce noise going back into the differential pair,  as well as provide a much smoother correction with abrupt volume changes by using the 2 stage input filter.  Opamp noise becomes a problem with such high gain used. I also used a very low noise 10 x battery battery preamp ahead of my CRO to confirm the initial improvements with my Preamp.

 Later improvements could only be verified by extended listening, as well as reports from other constructors of my design.

 The Offset Corrector can be seen in the bottom right hand corner of the Preamp/HA PCB

Class A HA Preamp PCB single.jpg

So to get back to whether you are hearing the improvement because of lower noise or something else... 

Don't you think that greater attenuation in the audible band in the revised circuit  should result in lower overall distortion from the power amplifier?  And so maybe this is the improvement you are hearing, not the lower noise?

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2 hours ago, sandyk said:

 

Not too many are likely to agree that we can hear a further reduction in distortion below the already very low level of < .0006% ¬¬

 

Is this the 0.0006% figure for the op amp?  The distortion I was talking about would be from the residual of the filtered power amplifier's waveform (audible frequencies), which is part of what is sent back to the amp for the purpose of offset correction.  In the original circuit (bottom image), I think this has a 90 degree phase shift in the audible band, which can't be good.  In the updated (top) circuit, the two pole input filter gives a 180 degree phase shift, so I would guess this and also the much greater attenuation by the input filter would give a very good improvement, much more than the lowering of the noise.  Correct me if my thinking is wrong; I am not very experienced in audio circuit design.

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