In Search Of Accurate Sound Reproduction: The Final Word!

[CuteStudio]

6 minutes ago, AJ Soundfield said:

You claimed 44.1kHz filtering can't be done

 

No I didn't, I said they didn't work: in the context of accurate sound reproduction (this thread)

They all have compromises which is why everyone started over and up sampling.

https://www.stereophile.com/features/106ringing/index.html

 

Whereas on the contrary 96kHz is far easier to build a filter for (and therefore a better sound).

 

I wonder if I misread  the title of the thread, I thought this was about accurate reproduction - if not, what's the point of the thread?

 

 

[mansr]

19 minutes ago, CuteStudio said:

Yes you are right, 2.05kHz if one is aiming for 20kHz. 0.05kHz was a math-typo.

 

I said:

 

"44.1kHz. No one can build a real filter that works for that"

 

Referring to the DAC anti-aliasing filter which has to be analog.

As I said, just about all DACs upsample digitally to simplify the analogue filter.

[AJ Soundfield]

8 minutes ago, CuteStudio said:

 

No I didn't, I said they didn't work: in the context of accurate sound reproduction

That's even worse, given that I just gave you a link by "HiRez" proponents no less, showing numerous studies falsifying your claim. That it was cited by Bob Stuart as evidence for the latest reincarnation, borders on the incredible.

http://journal.frontiersin.org/article/10.3389/fpsyg.2017.00093/full

 

Quote

https://www.stereophile.com/features/106ringing/index.html

 

Yes, another fantasy referencing Oohashis debunked nonsense about brainwaves showing "feeling better" due to inaudible HF (a word used often in the study I linked)

 

Quote

Whereas on the contrary 96kHz is far easier to build a filter for (and therefore a better sound).

Evidence please

[Jud]

14 minutes ago, CuteStudio said:

 

No I didn't, I said they didn't work: in the context of accurate sound reproduction (this thread)

They all have compromises which is why everyone started over and up sampling.

https://www.stereophile.com/features/106ringing/index.html

 

Whereas on the contrary 96kHz is far easier to build a filter for (and therefore a better sound).

 

I wonder if I misread  the title of the thread, I thought this was about accurate reproduction - if not, what's the point of the thread?

 

 

 

No one builds analog filters for 96kHz either.  They’re built for the output of SDM modulators, generally.

[CuteStudio]

Just now, AJ Soundfield said:

That's even worse, given that I just gave you a link by "HiRez" proponents no less, showing numerous studies falsifying your claim. That it was cited by Bob Stuart as evidence for the latest reincarnation, borders on the incredible.

http://journal.frontiersin.org/article/10.3389/fpsyg.2017.00093/full

 

 

Yes, another fantasy referencing Oohashis debunked nonsense about brainwaves showing "feeling better" due to inaudible HF (a word used often in the study I linked)

 

Evidence please

 

Thanks for the nice chat AJ, much appreciated.

[jabbr]

1 hour ago, Jud said:

 

No one builds analog filters for 96kHz either.  They’re built for the output of SDM modulators, generally.

Yeah or upsampled PCM ... boy the quality of Redbook is so good upsampled either to DSD or PCM I think that's at least 90% of the problem with 16/44 ... I can hear differences with higher res files but it's really hard to know how much of that isn't mastering etc as opposed to something intrinsically better about high res files. For example, my favorite Led Zep files are 16/44 -- @bdiament -- now he hears improvements with high res ... but ... doesn't AFAIK, listen using the upsampling software that we do.

[fas42]

16 minutes ago, jabbr said:

Yeah or upsampled PCM ... boy the quality of Redbook is so good upsampled either to DSD or PCM I think that's at least 90% of the problem with 16/44 ... I can hear differences with higher res files but it's really hard to know how much of that isn't mastering etc as opposed to something intrinsically better about high res files. For example, my favorite Led Zep files are 16/44 -- @bdiament -- now he hears improvements with high res ... but ... doesn't AFAIK, listen using the upsampling software that we do.

 

Agree about those Led Zep recordings - these blew my mind, back in the 1980's from CD; fabulous, fabulous stuff!

 

What upsampling does is allow the playback chain to have an easier time of getting the reproduction correct - particularly useful for "lower quality" gear - an analogy is using higher impedance speakers with ordinary amplifiers, so high drive currents are not required.

[audiventory]

On 18.06.2017 at 2:12 PM, AJ Soundfield said:

"We" who?

Somebody right now able to capture and restore soundfield of concert hall directly in human brain?

[AJ Soundfield]

15 minutes ago, audiventory said:

Somebody right now able to capture and restore soundfield of concert hall directly in human brain?

That was your contention, probably been watching too much Matrix. Individuals lifetime perception of audio and stored "reality" have those pesky ears in the way.

I'm sure most here would be ok with external soundfields like the PSR one I linked. YMMV.

[audiventory]

9 minutes ago, AJ Soundfield said:

been watching too much Matrix

I known about the subjective world perception (I don't know proper philosophical term in English) many years before the movie in my childhood. It is "slightly" older theory than the film

 

9 minutes ago, AJ Soundfield said:

I'm sure most here would be ok with external soundfields like the PSR one I linked.

I never sure. I know that all may be changed with time. Audibility threshold is most sophisticated thing in audio industry.

[CuteStudio]

16 hours ago, jabbr said:

Yeah or upsampled PCM ... boy the quality of Redbook is so good upsampled either to DSD or PCM I think that's at least 90% of the problem with 16/44 ... I can hear differences with higher res files but it's really hard to know how much of that isn't mastering etc as opposed to something intrinsically better about high res files. For example, my favorite Led Zep files are 16/44 -- @bdiament -- now he hears improvements with high res ... but ... doesn't AFAIK, listen using the upsampling software that we do.

 

Surely upsampling involves digital interpolation (estimation)  of the intermediate values between the original data points? Is result then still redbook at all?

 

The process of mastering to 44.1 and then upsampling at the playback end could be described as a form of lossy compression because essentially you are throwing away data to create the 44.1 waveform, and then estimating what those values may have been during the upsampling.

 

The need to do this appears to corroborate my initial (oddly controversial) contention that anti aliasing filters don't work (very well) on the 44.1kHz redbook standard.

 

I'm not sure why we don't just store the music at 96kHz anyway, negating the need for interpolation and upsampling, 44.1 was only chosen due to the technical limitations we had 30 years ago.

 

[Jud]

18 minutes ago, CuteStudio said:

I'm not sure why we don't just store the music at 96kHz anyway, negating the need for interpolation and upsampling, 44.1 was only chosen due to the technical limitations we had 30 years ago.

 

You wouldn't negate the "need" for upsampling unless you stored audio in the form that it's sent to the analog reconstruction filter in the vast majority of DACs, which would be a sigma-delta modulated format using MHz sample rates.

 

Of course the reason 44.1 is still used is because a sizable economic chunk of the music distribution chain is still CDs.  If you really wanted to store stuff in the format in which most of it winds up being used, that would be mp3 or AAC.

[AJ Soundfield]

29 minutes ago, CuteStudio said:

The need to do this appears to corroborate my initial (oddly controversial) contention that anti aliasing filters don't work (very well) on the 44.1kHz redbook standard.

Evidence please.

Assuming even for a moment you mean "audible", please present contrary Redbook filter data to this http://www.aes.org/e-lib/browse.cfm?elib=15249

[AJ Soundfield]

31 minutes ago, CuteStudio said:

I'm not sure why we don't just store the music at 96kHz anyway

That is how it's been stored for around 2 decades.

It just makes more sense to distribute at 44k due to file size and human hearing

[mansr]

19 minutes ago, CuteStudio said:

Surely upsampling involves digital interpolation (estimation)  of the intermediate values between the original data points? Is result then still redbook at all?

 

The process of mastering to 44.1 and then upsampling at the playback end could be described as a form of lossy compression because essentially you are throwing away data to create the 44.1 waveform,

This process is correctly termed band-limiting.

19 minutes ago, CuteStudio said:

and then estimating what those values may have been during the upsampling.

If the original recording had frequency content above 22.05 kHz, this is irretrievably lost, and upsampling does not involve guessing what it might have been. Rather, upsampling interpolates sample values such that the bandwidth is extended while preserving as closely as possible the frequencies in the input.

19 minutes ago, CuteStudio said:

The need to do this appears to corroborate my initial (oddly controversial) contention that anti aliasing filters don't work (very well) on the 44.1kHz redbook standard.

Digital filters work very well indeed. Analogue filters are trickier, but not impossible, which (I'm starting to feel like a parrot) is why DACs upsample digitally before applying a simpler analogue low-pass filter. Nothing is remotely controversial here.

[monteverdi]

My take (in agreement with previous poster) on the major limitations on accurate sound production is in and out i.e. recording and speaker/room.

 

Even if format (analog/digital and which file format) maters it is very secondary to the recording and post recording processing. Especially multiple microphones and the generation of an artificial soundstage in mixing can not produce an accurate reproduction of acoustic music. A lot of releases from the music industry reminds me of the attitude of  industrial food production. (May be times are changing when Amazon is buying Whole Foods).That does not mean I can not enjoy electronic music but that production scheme should not applied to everything.

 

Loudspeakers got incrementally better but at a much slower speed than the rest of the audio equipment, especially digital. Real improvements and more radical solutions are scarce. There are lot of "flagship speakers" of high complexity at ridiculous prices. Instead of obsessing about equipment one should invest in better rooms. I see on the web a lot of pictures of rooms full of expensive toys but no consideration to room acoustics. At least there should be left right symmetry for proper imaging and not a window on one side and a bookshelf on the opposite.

A lot of room treatment is focused absorption but a lot of absorbers have very low efficiency at low frequency and therefore color the acoustic impression. Reflections are not negative if they have enough time delay and are attenuated but they are pretty much unavoidable from the floor ( a rug works only at high frequency). Bringing architectural consideration and room acoustics together requires a lot of thought and planning (presently trying to design my own living/listening room).

 

[Jud]

1 minute ago, mansr said:

Digital filters work very well indeed. Analogue filters are trickier, but not impossible, which (I'm starting to feel like a parrot) is why DACs upsample digitally before applying a simpler analogue low-pass filter. Nothing is remotely controversial here.

 

Which is why I'm wondering why folks feel the need to continue to argue pro or con over a situation (44.1kHz input to a final analog reconstruction filter) that almost never happens any more.

 

Upsampling became standard very quickly, and sigma-delta modulation became standard not too long after that, because they made the final filtering easier and (key word here) cheaper.  Final reconstruction filtering wasn't impossible with 44.1kHz input (as mansr says), but it was more expensive to do well, and that was the death of it.

[AJ Soundfield]

4 minutes ago, monteverdi said:

the major limitations on accurate sound production is

This http://www.aes.org/e-lib/browse.cfm?elib=9136

Not idiotic sample rates and word length with 2 channels.

Nothing to "reproduce" when it isn't there!

[monteverdi]

I was listening to a presentation of Weiss at the last High End in Munich and they showed a prototype system using an extra large sound-bar type loudspeaker and processing based on HRTF of binaural recordings. Was very impressive as long on sits in the sweet spot. That restriction and the lack of binaural recordings makes it difficult for me to get too excited.

[Ralf11]

20 hours ago, fas42 said:

 

What upsampling does is allow the playback chain to have an easier time of getting the reproduction correct - particularly useful for "lower quality" gear - an analogy is using higher impedance speakers with ordinary amplifiers, so high drive currents are not required.

 

 

I'm not sure what that means.  I thought that what upsampling does is allow the use of more gentle analog filter slopes, and that any interpolation errors are less of an issue for SQ than the ability to avoid 'brickwall' filters.

 

I also sometimes hear that integer value upsampling should be used, but it is not clear ot me why that is advocated, tho it makes the FFT math easier.

[esldude]

2 minutes ago, Ralf11 said:

 

 

I'm not sure what that means.  I thought that what upsampling does is allow the use of more gentle analog filter slopes, and that any interpolation errors are less of an issue for SQ than the ability to avoid 'brickwall' filters.

 

I also sometimes hear that integer value upsampling should be used, but it is not clear ot me why that is advocated, tho it makes the FFT math easier.

 

Actually some people say, (among then John Siau of Benchmark), that non-interger upsampling is slightly superior.

[esldude]

3 hours ago, AJ Soundfield said:

This http://www.aes.org/e-lib/browse.cfm?elib=9136

Not idiotic sample rates and word length with 2 channels.

Nothing to "reproduce" when it isn't there!

Yes, but where can we buy software or hardware to implement this.

 

The description sounded very worthwhile, yet is owned by the people JJ worked for and to my knowledge has not been licensed by anyone.  Is it available somewhere?  I would like to try it out if so.

[mansr]

2 hours ago, Ralf11 said:

I also sometimes hear that integer value upsampling should be used, but it is not clear ot me why that is advocated, tho it makes the FFT math easier.

There is no difference in accuracy between integer and non-integer ratios.

[jabbr]

3 hours ago, Ralf11 said:

I also sometimes hear that integer value upsampling should be used, but it is not clear ot me why that is advocated, tho it makes the FFT math easier.

Probably the DAC only needs one master clock and rates could be upsampled to a 2^n multiple of this (typically n is negative)

but DACs usually have two clocks, one for the 44100 hz family and the other for 48000. It's much easier for the DAC to generate the BCLK by division. 

[mansr]

13 minutes ago, jabbr said:

Probably the DAC only needs one master clock and rates could be upsampled to a 2^n multiple of this (typically n is negative)

but DACs usually have two clocks, one for the 44100 hz family and the other for 48000. It's much easier for the DAC to generate the BCLK by division. 

Yes, it's easier to make a chip where all clocks have small integer ratios. That is why power of two multiples are common. If done in software, these constraints don't apply, and arbitrary ratios can be used.