How many bits, how fast, just how much resolution is enough?

[semente]

@elsdude is correct, if the system is only responsive to 20khz then 44 khz is enough to sample the relevant transients. Conversely, if the system is sensitive to faster transients then it is *by definition* responsive to > 20 khz.

 

I'm not sure the 20khz limit is set in stone. That could be the limit for detecting isolated tones but since the (human auditory) system is highly nonlinear, that doesn't mean the entire system's response is limited to 20khz.

 

Consider the retina: although not sensitive to infra-red, the skin is :-)

 

It is possible that 30 khz frequencies are interpreted as "irritating" or other nonlinearities such as the response of amplifiers/electronics to "ultrasonics". Consider why we need to filter out > 20khz from DAC outputs: the system itself is sensitive to ultrasonics. These ultrasonic effects on electronics (that manifest themselves in the audible range) are examples of nonlinearities in the electronics -- similarly the auditory system itself may have (other) nonlinearities.

 

As they say, when you assume ...

 

In any case we really aren't doing science here so folks should judge for themselves. I find that DSD256 goes well with scotch.

 

Let's not forget the oil can resonance of many hard-domed tweeters which once triggered may very well have repercussions further down in the audible range:

 

 

Focal Aria 936, cumulative spectral-decay plot on tweeter axis at 50" (0.15ms risetime).

Read more at Focal Aria 936 loudspeaker Measurements | Stereophile.com

 

R

[BlueSkyy]

No doubt about that. But it isn't all about frequency response. There are other criteria that define fidelity. <------- What he said.

That's sage information.

[mansr]

No doubt about that. But it isn't all about frequency response. There are other criteria that define fidelity.

 

Yes, there is also non-linear distortion. The only distortion inherent in PCM audio is that caused by quantisation, and this is readily addressed with dither.

[mayhem13]

Of course it is. Imagine two equivalent 10 KHz pure analog sinusoidal tones, one started 100 microseconds before the other. A digital (PCM) 44.1/48 KHz sampled recording of this will sound different from a higher resolution digital recording..

 

It's important to look at these concerns in relative terms and conditions. In the case of your example, given the wavelength of 1.35" at 10khz, if your head at the time of listening or your speakers in the horizontal plane are off axis 1 degree or more, you've already exceeded your 100 microsecond time smear by a magnitude of 3 or greater.......and this assumes laboratory consistency for everything else in the signal and environmental chain.

 

I agree, you can make an argument for anything, but relevant arguements are few and far between. If i take 2 samples of a high end tweeter, install them on the same fixed test baffle without changing anything in the signal chain or testing environment the two responses, CSD, distortion sweeps, phase measure or impedance will NEVER measure exactly the same. Add humanistic variables like blood pressure and glucose levels and the waters get even murkier for your position. Coefficients of significance.

[Miska]

The only distortion inherent in PCM audio is that caused by quantisation, and this is readily addressed with dither.

 

Plus images, which is fully correlated distortion... And of course all the non-linearity distortions of a R2R ladder DAC.

[Miska]

I would be more convinced if someone had rational explanations for what point it sounds so good no improvement can be detected with higher sample rates. So far it is just higher is better as if the improvement can be extended forever with more more more.

 

I have posted quite a number of measurement results to show what are the differences... What else is needed?

[Miska]

Let's not forget the oil can resonance of many hard-domed tweeters which once triggered may very well have repercussions further down in the audible range:

 

Yes, that is particularly exaggerated by leaky oversampling filters with RedBook sources triggering the resonance all the time.

 

As I've said, the sound gives me feeling of being in dentist chair. It's the "CD sound of 80's". It hurts... Some people think it sounds "accurate", to me it sounds hissy-messy.

[mansr]

Plus images, which is fully correlated distortion... And of course all the non-linearity distortions of a R2R ladder DAC.

 

DAC imperfections are not part of the data format.

[Miska]

DAC imperfections are not part of the data format.

 

Sure not, same applies to any data format, be it PCM or DSD. Imperfections come from the algorithms involved in generating and dealing with the data. And from the converters, A/D and D/A. Some are better, some are worse...

 

What matters in the end is what comes out of the DAC in analog form. That's the ultimate measure.

[Miska]

It's important to look at these concerns in relative terms and conditions. In the case of your example, given the wavelength of 1.35" at 10khz, if your head at the time of listening or your speakers in the horizontal plane are off axis 1 degree or more, you've already exceeded your 100 microsecond time smear by a magnitude of 3 or greater.......and this assumes laboratory consistency for everything else in the signal and environmental chain.

 

Typical time domain spread of a RedBook oversampling filter is around 1 millisecond.

[sdolezalek]

I'm assuming for this post that 16/44 captures all of the frequency range and transient information we need (you may disagree, but make that assumption for a moment). Then the real reason for hi-res (or upsampling) is all about the digital-to-analog conversion process and the benefits we get by moving filters and other conversion artifacts as far above 20kHz as we can.

 

That results in

a) our wanting to feed our DAC hi-res (call it DSD128 or higher or DXD) information (irregardless of whether it started as 16/44 that we upsampled or an actual DSD128 or DSD512 or DXD file);

b) then applying less aggressive filters (so as to avoid ringing and other artifacts in the 20Hz-20kHz range only); and

c) feeding all equipment downstream of the DAC considerable content above 20kHz (unless the DAC itself introduces a 20kHz cliff filter).

 

If we are doing a,b and c, then how important is it for us to know how each piece of downstream equipment will interact with those higher than 20kHz frequencies (i.e. pre-amp, amp and tweeters)?

 

Some have made the case for having pre-amps, amps and speakers that are linear out to 40kHz or more such that feeding them information not brickwalled at 20kHz doesn't introduce new problems that we were trying to solve with the hi-res and gentle filters solution at the DAC. Is it therefore possible that the reason some of us do/don't prefer (or hear) the benefits of hi-res is the limitations of the downstream equipment?

[tailspn]

Is it therefore possible that the reason some of us do/don't prefer (or hear) the benefits of hi-res is the limitations of the downstream equipment?

 

That does suggest one reason for the perception differences among listeners to DSD delivered recordings using different bit rates. The above 20KHz energy content of a DSD256 delivery to a DAC is much lower than a DSD64 for the same 20 -20KHz music spectral content. How downstream equipment, especially amplifiers, react to that energy (uncorrelated white noise) has always been an issue.

[esldude]

I have posted quite a number of measurement results to show what are the differences... What else is needed?

I seem to have missed where you have the specs for a sample rate filter combination so good no further audible improvement is possible by increasing sample rate or improving the filter. One might make a better measured result beyond this, but hear not the improvement. Care to tell us what it is?

 

That way there is a target instead of this mindless idea more is better ad infinitum.

 

Sent from my Nexus 6P using Computer Audiophile mobile app

[Jud]

I'm assuming for this post that 16/44 captures all of the frequency range and transient information we need (you may disagree, but make that assumption for a moment). Then the real reason for hi-res (or upsampling) is all about the digital-to-analog conversion process and the benefits we get by moving filters and other conversion artifacts as far above 20kHz as we can.

 

That results in

a) our wanting to feed our DAC hi-res (call it DSD128 or higher or DXD) information (irregardless of whether it started as 16/44 that we upsampled or an actual DSD128 or DSD512 or DXD file);

b) then applying less aggressive filters (so as to avoid ringing and other artifacts in the 20Hz-20kHz range only); and

c) feeding all equipment downstream of the DAC considerable content above 20kHz (unless the DAC itself introduces a 20kHz cliff filter).

 

If we are doing a,b and c, then how important is it for us to know how each piece of downstream equipment will interact with those higher than 20kHz frequencies (i.e. pre-amp, amp and tweeters)?

 

Some have made the case for having pre-amps, amps and speakers that are linear out to 40kHz or more such that feeding them information not brickwalled at 20kHz doesn't introduce new problems that we were trying to solve with the hi-res and gentle filters solution at the DAC. Is it therefore possible that the reason some of us do/don't prefer (or hear) the benefits of hi-res is the limitations of the downstream equipment?

 

 

You're missing one piece, which is that moving the sample rate higher permits exclusion of near-20KHz ultrasonics without a "brickwall" filter. For DSD, ultrasonic noise is moved further from the audible range.

[Jud]

Some graphs of noise vs. frequency for lower and higher DSD rates (audibility of course is another matter):

 

http://www.computeraudiophile.com/f11-software/audirvana-plus-2-multi-channel-sound-5-surround-channels-and-1-lfe-low-frequency-effects-channel-pulse-code-modulation-direct-stream-digital-conversion-28428/index30.html#post594841

 

http://www.computeraudiophile.com/f11-software/audirvana-plus-2-multi-channel-sound-5-surround-channels-and-1-lfe-low-frequency-effects-channel-pulse-code-modulation-direct-stream-digital-conversion-28428/index30.html#post594932

 

http://www.computeraudiophile.com/f11-software/audirvana-plus-2-multi-channel-sound-5-surround-channels-and-1-lfe-low-frequency-effects-channel-pulse-code-modulation-direct-stream-digital-conversion-28428/index30.html#post598722

 

Ultrasonic (test) signals causing distortion in the audible range with a particular DAC (though any effect with normal music is felt to be unlikely by the thread participants):

 

http://www.computeraudiophile.com/f30-abbingdon-music-research-ifi-audio-sponsored/ifi-nano-intermodulation-distortion-27987/#post523544

[Miska]

I seem to have missed where you have the specs for a sample rate filter combination so good no further audible improvement is possible by increasing sample rate or improving the filter. One might make a better measured result beyond this, but hear not the improvement. Care to tell us what it is?

 

If you think you don't hear the measurable improvements, including improvements in the audio band, it's your opinion. However, you cannot claim that nobody else in this world is able to hear it.

 

That way there is a target instead of this mindless idea more is better ad infinitum.

 

As long as measurable (and/or audible) signal fidelity keeps improving by improving implementation, I will keep doing it ad infinitum. I have no reason not to. So far I've been also hearing improvements, but not going to claim anything about anybody else's hearing or non hearing. That's something they need to decide on their own.

 

If improvements become unmeasurable, there just needs to be better measurements. Luckily measurement equipment also keeps improving at steady pace.

 

My target is beyond what is possible, so the target will be never reached. That's how I like it.

[sdolezalek]

You're missing one piece, which is that moving the sample rate higher permits exclusion of near-20KHz ultrasonics without a "brickwall" filter. For DSD, ultrasonic noise is moved further from the audible range.

 

Jud: Thanks for adding that. Could you explain a bit more? What causes the "exclusion of the near 20kHz ultrasonics?" I assume that is the same difference that tailspn is referring to in terms of less >20kHz content in DSD256 than DSD64?

 

(Sorry, posted too quickly; the graphs you cross reference in your post above, do explain it. Thanks)

[Miska]

Some have made the case for having pre-amps, amps and speakers that are linear out to 40kHz or more such that feeding them information not brickwalled at 20kHz doesn't introduce new problems that we were trying to solve with the hi-res and gentle filters solution at the DAC. Is it therefore possible that the reason some of us do/don't prefer (or hear) the benefits of hi-res is the limitations of the downstream equipment?

 

Likely yes... I have paid particular attention to use equipment that can cleanly reproduce at least to ~50 kHz.

 

Some of the improvements are audible also without such equipment, for example improvements in intermodulation and time domain behavior. Also in many cases jitter and level linearity performance improves.

[esldude]

If you think you don't hear the measurable improvements, including improvements in the audio band, it's your opinion. However, you cannot claim that nobody else in this world is able to hear it.

 

 

 

As long as measurable (and/or audible) signal fidelity keeps improving by improving implementation, I will keep doing it ad infinitum. I have no reason not to. So far I've been also hearing improvements, but not going to claim anything about anybody else's hearing or non hearing. That's something they need to decide on their own.

 

If improvements become unmeasurable, there just needs to be better measurements. Luckily measurement equipment also keeps improving at steady pace.

 

My target is beyond what is possible, so the target will be never reached. That's how I like it.

 

So you don't have an answer to my question. ad infinitum is your approach.

 

Thank you for your response.

[mansr]

So you don't have an answer to my question. ad infinitum is your approach.

 

No surprise. His business depends on it.

[Jud]

Well this was written in 2004

 

But as you point out in a part of your post I snipped, he said the same in 2012, and as far as I know continues to maintain the same position. See, e.g., http://www.lavryengineering.com/pdfs/lavry-white-paper-the_optimal_sample_rate_for_quality_audio.pdf

 

The idea to keep pushing sample rates up both in the operation of delta-sigma designs and in the PCM or DSD available rates is an easy selling point. Just like megapixels for cameras. I would be more convinced if someone had rational explanations for what point it sounds so good no improvement can be detected with higher sample rates. So far it is just higher is better as if the improvement can be extended forever with more more more.

Yes, we will always have specsmanship in the marketing of these products. On the other hand, I don't see *current* support for Lavry's position that anything over a 96KHz sample rate leads to worse results.

[Jud]

I'm not sure this from Miska -

 

As long as measurable (and/or audible) signal fidelity keeps improving by improving implementation, I will keep doing it ad infinitum. I have no reason not to. So far I've been also hearing improvements, but not going to claim anything about anybody else's hearing or non hearing. That's something they need to decide on their own.

[emphasis added]

 

- really deserved the rather negative responses to it. Seems pretty reasonable to me.

[mayhem13]

Typical time domain spread of a RedBook oversampling filter is around 1 millisecond.

 

Only working with his example buddy......but even for a millisecond (.001) there's still no relevance unless ones head is clamped in a vise of chair positioned in a measured equilateral triangle in an anechoic room of equandestant dimensions.

 

More of a matter of preference for me but I tend not to listen under those conditions! Lol

I like to move around a bit, ya know?

 

God help us all if the culinary world develops the same audiophobia criteria where everything matters! Who wants a lab technician instead of a Michelin chef?

[Miska]

No surprise. His business depends on it.

 

Luckily I don't depend on my business. (it is not what I do for living)

 

But overall, how is that different from any other audiophile development? Maybe your profile picture explains your opinion in the most clear way. I don't share that opinion though.

 

I was doing HQPlayer for myself long before I started selling licenses for it, just thought that since I have something I could also make it available to others while I keep improving it. My business was started for doing audio measurement/analysis systems... Lot of the same algorithms just happen to fit for HQPlayer purpose too.

[mayhem13]

Likely yes... I have paid particular attention to use equipment that can cleanly reproduce at least to ~50 kHz.

 

Some of the improvements are audible also without such equipment, for example improvements in intermodulation and time domain behavior. Also in many cases jitter and level linearity performance improves.

 

Again.......I've figured after a few years of this some of you would have stopped trying to separate time from frequency in an effort to incorrectly attribute suspected audible differences to this mystical place. Can't have one without the other fellas! If timing changes, so does the response at the listening position.