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Author Topic: 44k vs. 96k - a new discovery  (Read 15709 times)

dvuckovic

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Re: 44k vs. 96k - a new discovery
« Reply #15 on: October 24, 2004, 11:41:38 PM »

Nika Aldrich wrote on Sun, 24 October 2004 21:08

Barry Hufker wrote on Sat, 23 October 2004 19:50






I thinik the chip you may be speaking about is the AKM AK5394A.  It is indeed a very good chip.

Nika.


Correct me if I'm wrong, but this could as well be the new Texas Instruments chip.

As far as I know it it supposed to be much better than the trusty-old AKM found in so much audio gear(originally intended for the A/D in medical instruments)
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Nika Aldrich

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Re: 44k vs. 96k - a new discovery
« Reply #16 on: October 24, 2004, 11:44:10 PM »

dvuckovic wrote on Mon, 25 October 2004 04:41

Nika Aldrich wrote on Sun, 24 October 2004 21:08

Barry Hufker wrote on Sat, 23 October 2004 19:50






I thinik the chip you may be speaking about is the AKM AK5394A.  It is indeed a very good chip.

Nika.


Correct me if I'm wrong, but this could as well be the new Texas Instruments chip.

As far as I know it it supposed to be much better than the trusty-old AKM found in so much audio gear(originally intended for the A/D in medical instruments)


The 5394 is a relatively new chip that was designed exclusively for audio purposes, as is indicated by its filtering and noiseshaping curves.  AKM has had several chips in the past, but the 5394 is getting a tremendous amount of respect in design circles.

The other three companies (BB, AD, and Crystal) all have competing hardware and new chip advances as well, but I believe the quote being referred to is the AKM chip.

Nika.
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Joe Bryan

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Re: 44k vs. 96k - a new discovery
« Reply #17 on: October 25, 2004, 01:38:28 AM »

Nika Aldrich wrote on Sun, 24 October 2004 11:58

Joe Bryan wrote on Sun, 24 October 2004 07:45

Specifically, well designed DSP filters will sound better at higher sampling rates because their responses can more closely approach the ideal analog response (in technical terms, the s-plane to Z-plane warping is more linear).


Not all filters perform better at higher sampling rates.  Some filters have greater accumulative error by using excess taps to perform the process.


Noise accumulation is a function of the rounding and/or dither modes used during accumulation in the filter, and as I said, any well designed filter will take this into account. Noise buildup is not as big of an issue as Z-plane warping, which causes increasing deviations from analog (s-plane) transfer functions in the upper octave(s).

In many cases, this makes it impossible to reproduce the exact transfer function without upsampling. For example, it's impossible to recreate the exact amplitude and phase characteristics of most high-order (>2) analog filters in digital without upsampling. A perfect example is the Pultec EQ. Any DSP process that claims to be a Pultec but doesn't upsample isn't matching the response correctly.

Nika Aldrich wrote on Sun, 24 October 2004 11:58

Quote:

Futhermore, at higher sampling rates, analog anti-aliasing and anti-imaging filter requirements are less stringent (allowing less phase shift in the main audio band),


Not true.  Almost all anti-aliasing and anti-imaging filters that you are speaking of are digital and linear phase, so no phase shift issues are improved by higher sampling rates.  The analog filters in any oversampling filter these days have phase shift so far below our audible phase discernability that this is clearly not a factor.  Further, most high sample rate converters use the same analog anti-aliasing/anti-imaging filters as their lower sample rate brethren.

Indeed, the anti-aliasing and anti-imaging filters are less stringent at the higher sampling rates, but this only pertains to what you call "poorly designed filters."  Not all filters have to suffer from the problems of the poorly designed ones, and the better designed ones will perform the same at the lower rate as at the higher rate.


I was referring to the analog filters, not the digital filters.

The digital filters only do part of the work. For a typical sigma-delta ADC, the analog anti-aliasing filter is trivial because it only needs to attenuate signals above the MHz range.

However, the analog anti-imaging filter for a DAC is not trivial. In today's sigma-delta converters, there is a lot of out-of-band energy that cannot be removed by the digital filter because it's ahead of the signal-delta DAC's modulator. The modulator's noise shaping shifts the quantization noise out of the main audio band, but this energy can only be removed by the analog filter.

It is this filter that benefits the most from higher sampling rates. The higher transition band provided by higher sampling rates allows for much less phase shift in the primary audio band while attenuating the high-freq energy that wreaks havock in the analog output amps. This has a major impact on sound quality, especially transparency and transient response.

Nika Aldrich wrote on Sun, 24 October 2004 11:58

Quote:

and system latency can be reduced so digital processors can be used in real-time monitoring situations (especially vocal tracking).


For the most part system latency is improved with higher sample rates, though there are some situations where latency takes a hit as sample rates increase.

Nika.


I can't think of any examples where this is true unless you're referring to up/down SRC when the processing is oversampled but not the analog conversion, could you provide some?

-Joe
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Nika Aldrich

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Re: 44k vs. 96k - a new discovery
« Reply #18 on: October 26, 2004, 11:29:02 AM »

Joe Bryan wrote on Mon, 25 October 2004 06:38

In many cases, this makes it impossible to reproduce the exact transfer function without upsampling. For example, it's impossible to recreate the exact amplitude and phase characteristics of most high-order (>2) analog filters in digital without upsampling. A perfect example is the Pultec EQ. Any DSP process that claims to be a Pultec but doesn't upsample isn't matching the response correctly.


There are a lot of EQs out there that successfully do proper DSP without upsampling.  The Sony Oxford is a classic example.

Quote:

I was referring to the analog filters, not the digital filters.

The digital filters only do part of the work. For a typical sigma-delta ADC, the analog anti-aliasing filter is trivial because it only needs to attenuate signals above the MHz range.

However, the analog anti-imaging filter for a DAC is not trivial. In today's sigma-delta converters, there is a lot of out-of-band energy that cannot be removed by the digital filter because it's ahead of the signal-delta DAC's modulator. The modulator's noise shaping shifts the quantization noise out of the main audio band, but this energy can only be removed by the analog filter.


Sure, but the roll-off of that analog filter is so far out of band that increasing it further is unnecessary with respect to phase shift in the audible range.

Quote:

It is this filter that benefits the most from higher sampling rates. The higher transition band provided by higher sampling rates allows for much less phase shift in the primary audio band while attenuating the high-freq energy that wreaks havock in the analog output amps. This has a major impact on sound quality, especially transparency and transient response.


On a typical DSM based DAC the oversampling brings the excess noise so far above the audible range that a very gentle filter can be used, just like on the A/D side.  I see no grounds for claiming that making the filter even more gentle is better on audio band material - especially if the current system provides no audible phase distortion because the filter is already gentle enough.

Quote:

I can't think of any examples where this is true unless you're referring to up/down SRC when the processing is oversampled but not the analog conversion, could you provide some?


There are plugins that, when run at higher sampling rates, have to down sample in order to do the processing and then upsample at the end of the process.  

Nika.
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danlavry

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Re: 44k vs. 96k - a new discovery
« Reply #19 on: October 26, 2004, 03:33:22 PM »

Barry Hufker wrote on Sun, 24 October 2004 04:45

Dan,

I head the 4-year undergraduate program in Audio Production at Webster University.  We invited Rupert Neve as a guest speaker.  I say this only to give you the circumstances under which Mr. Neve came to our campus this past spring to speak.  It was at that time (April or May of this year, I forget exactly), that Mr. Neve made the statement I attributed to him.  He even said this new chip was just coming out.

I was a witness to this statement, but this is not my area of expertise so I can't clarify it further.  But Mr. Neve spoke of it as an immediate (at that moment) development.

Barry



Hello Barry,

There is probably some misunderstanding here of some sort.
Again, the -1dB at 20KHz problem is an old thing from an anti aliasing filter standpoint. The new converters are way passed such problems.

Technically speaking, the -1dB at 20KHz when operating at say 44.1KHz, is a very real issue, but is due to the digital decimation filter, not the analog anti aliasing filter.

Again, the reason for my statement is due to the increased rate operation of the front end (the modulator).

I can not tell the reason for the misunderstanding. I do however stand behind my statements, which are just technical facts.

BR
Dan Lavry

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Barry Hufker

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Re: 44k vs. 96k - a new discovery
« Reply #20 on: October 26, 2004, 05:41:43 PM »

Dan,

I am not an expert on this subject by any means and I don't dispute what you have said.  There probably is a misunderstanding on my part as to the full meaning of Mr. Neve's statement, but I think Nika summarized pretty well what I understood Mr. Neve to mean.

I can't elaborate further or else I would.  And I don't want to misrepresent Mr. Neve.  So maybe we'll have to let it go with the ground we've already covered.

Barry
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Joe Bryan

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Re: 44k vs. 96k - a new discovery
« Reply #21 on: October 27, 2004, 02:11:42 AM »

Nika Aldrich wrote on Tue, 26 October 2004 08:29

Joe Bryan wrote on Mon, 25 October 2004 06:38

In many cases, this makes it impossible to reproduce the exact transfer function without upsampling. For example, it's impossible to recreate the exact amplitude and phase characteristics of most high-order (>2) analog filters in digital without upsampling. A perfect example is the Pultec EQ. Any DSP process that claims to be a Pultec but doesn't upsample isn't matching the response correctly.


There are a lot of EQs out there that successfully do proper DSP without upsampling.  The Sony Oxford is a classic example.


Yes, there are plenty of EQs that can produce good filter responses without upsampling, but that does not mean they can produce every desireable filter response. As I said, there are analog filters that cannot be reproduced digitally without upsampling.

Nika Aldrich

Quote:

I was referring to the analog filters, not the digital filters.

The digital filters only do part of the work. For a typical sigma-delta ADC, the analog anti-aliasing filter is trivial because it only needs to attenuate signals above the MHz range.

However, the analog anti-imaging filter for a DAC is not trivial. In today's sigma-delta converters, there is a lot of out-of-band energy that cannot be removed by the digital filter because it's ahead of the signal-delta DAC's modulator. The modulator's noise shaping shifts the quantization noise out of the main audio band, but this energy can only be removed by the analog filter.


Sure, but the roll-off of that analog filter is so far out of band that increasing it further is unnecessary with respect to phase shift in the audible range.


It's not "so far out" as you say. The one plot you won't see on any of the DAC spec sheets is the out of band noise caused by the modulator's noise shaping. This rises quite rapidly above the digital filter's stop band, and requires some non trivial analog filtering to remove it. The analog filter must be set as low as possible to remove this noise, and this requires juggling greater in-band phase error vs. greater out-of-band noise levels.

Nika Aldrich

Quote:

It is this filter that benefits the most from higher sampling rates. The higher transition band provided by higher sampling rates allows for much less phase shift in the primary audio band while attenuating the high-freq energy that wreaks havock in the analog output amps. This has a major impact on sound quality, especially transparency and transient response.


On a typical DSM based DAC the oversampling brings the excess noise so far above the audible range that a very gentle filter can be used, just like on the A/D side.  I see no grounds for claiming that making the filter even more gentle is better on audio band material - especially if the current system provides no audible phase distortion because the filter is already gentle enough.


See above. Unless my ears and test equipment have been deceiving me all these years, it does make a difference.

Nika Aldrich

Quote:

I can't think of any examples where this is true unless you're referring to up/down SRC when the processing is oversampled but not the analog conversion, could you provide some?


There are plugins that, when run at higher sampling rates, have to down sample in order to do the processing and then upsample at the end of the process.


Ah, yes. Forgot about those. We don't usually have to deal with that problem because we still haven't run out of cycles on our DSP. Wink

Sometimes downsampling is done when the process doesn't benefit from the higher audio bandwidth, and the excess cycles are unneccessary. Digital reverb is one common example, and because of the time-smearing nature of reverb, the added delays from the SRC are trivial. This type of reverb would still benefit from lower-latency, high sample rate processing in a monitoring environment as long as the direct path wasn't downsampled.

Another example is decimated linear phase filters, which have high intrinsic delays (like any linear phase process). These would never be used in a monitoring situation to begin with.

The only other case where downsampling is necessary is when a process can't run at full speed without overloading the system. No algo designer wants to do this, they're forced to do it to get the process to run, even if it compromises quality.

-Joe
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ted nightshade

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Re: 44k vs. 96k - a new discovery
« Reply #22 on: November 12, 2004, 07:55:48 PM »

Since Ivo is using the Lavry Blue, maybe Dan can explain just what it is about that specific design that causes the difference in sound between 44.1 and 96 in that specific instance?

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bobkatz

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Re: 44k vs. 96k - a new discovery
« Reply #23 on: November 12, 2004, 08:04:49 PM »

ted nightshade wrote on Fri, 12 November 2004 19:55

Since Ivo is using the Lavry Blue, maybe Dan can explain just what it is about that specific design that causes the difference in sound between 44.1 and 96 in that specific instance?




Different filters yield different sound.

BK
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Bob Olhsson

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Re: 44k vs. 96k - a new discovery
« Reply #24 on: November 13, 2004, 10:25:14 AM »

I wouldn't assume this must be the new A to D converter rather than something downstream that his previous converter was masking.

Schallfeldnebel

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Re: 44k vs. 96k - a new discovery
« Reply #25 on: November 13, 2004, 11:23:31 AM »

Was the AK 5394A not the first chip with 192K possibility? Genex uses this chip in their GXA 85 AD converter for the PCM side, not for the DSD side. This is a multibit delta sigma chip. Weiss engineering uses this chip in the new ADC 2.

The chip Mr. Neve spoke about in his lecture, probably is a cousin from the Crystal 5397, which is used by DAD, Digital Audio Denmark in their 2402 and 2408 converters, and this converter chip is not new at all, it came together with the wellknown Crystal 5396 which Daniel Weiss used for a very short moment in the ADC1 mark 2.

Both chips get extremely hot. The 5397 is a chip, and DAD can explain it better themselves, see www.digitalaudio.dk ,which uses different filtering, and therefore reduces A.I.D., which stands for aliasing intermodulaton distortion. Because the chip gets so hot, I heard the DAD boys from Denmark speak about, another manufacturer, (Texas instruments ????) was interested in producing such chip, using less power and have a lower heat dissipation.

Maybe that is the chip Mr. Neve was speaking about.

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Brandon Schexnayder

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Re: 44k vs. 96k - a new discovery
« Reply #26 on: November 15, 2004, 05:03:15 PM »

Can I just ask a very naive question...as I am still quite new to the recording world:

Is it possible that there may be a much more basic reason for a subjective preference at 44kHz than the converters-- for instance an ingrained preference for audio at this sampling rate, simply due to what we have become so accustomed to?

I don't mean to negate that technology may have been substandard, I just genuinely wonder if there is more of an X factor at play here.
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danlavry

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Re: 44k vs. 96k - a new discovery
« Reply #27 on: November 15, 2004, 07:43:14 PM »

ted nightshade wrote on Sat, 13 November 2004 00:55

Since Ivo is using the Lavry Blue, maybe Dan can explain just what it is about that specific design that causes the difference in sound between 44.1 and 96 in that specific instance?




1. It is NOT the analog. It is NOT the clocks.

2. The biggest single difference in the AD (44.1KHz to 96KHz) is the DIGITAL decimation filter.

3. But some of the difference may also be in the DA side. At the DA the biggest difference is the DIGITAL up sampling filter.

Best Regards
Dan Lavry
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ted nightshade

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Re: 44k vs. 96k - a new discovery
« Reply #28 on: November 16, 2004, 10:35:48 AM »

Thanks Dan! You couldn't be clearer.
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Nika Aldrich

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Re: 44k vs. 96k - a new discovery
« Reply #29 on: November 16, 2004, 10:41:41 AM »

Schexnayder wrote on Mon, 15 November 2004 17:03

Is it possible that there may be a much more basic reason for a subjective preference at 44kHz than the converters-- for instance an ingrained preference for audio at this sampling rate, simply due to what we have become so accustomed to?


First we have to ascertain whether or not there is truly a difference in the audible results of the sampling rates.  So far everything is point at the audible difference being a manifestation of implementation - that if we design the circuits "right" then the difference goes away.  If that is the case then there would be no "preference."

Make sense?

Nika
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