Discussion:
Theoretical benefits to hi-res if only 20kHz signal?
Archimago
2012-02-06 17:19:32 UTC
Permalink
Hi all,
It's one of those questions I don't recall much discussion about.

If I'm recording some audio in PCM that contains no frequency >20kHz,
is there any benefit to recording this at 88kHz or more? That is, does
sampling theory suggest that the improved temporal resolution will add
anything once the signal is reconstructed in the DAC?

I've always assumed that the improved temporal resolution should give
us better transient response (ie. the slam of the kick drum might just
arrive slightly quicker and more precisely, whether the ear/brain
detects it is another matter). I'm thinking it might not be so obvious
once we go through the science of it all...

Regards,
Arch
--
Archimago
------------------------------------------------------------------------
Archimago's Profile: http://forums.slimdevices.com/member.php?userid=2207
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Phil Leigh
2012-02-06 18:32:59 UTC
Permalink
Post by Archimago
Hi all,
It's one of those questions I don't recall much discussion about. It
comes about because of music like Kent Poon's Audiophile Jazz Prologue
III (where the hi-res material looks like upsampled 44/48! yet he
claims it was natively recorded at hi-res but presumably a filter added
in postprocessing).
If I'm recording some audio in PCM that contains no frequency >20kHz,
is there any benefit to recording this at 88kHz or more? That is, does
sampling theory suggest that the improved temporal resolution will add
anything once the signal is reconstructed in the DAC?
I've always assumed that the improved temporal resolution should give
us better transient response (ie. the slam of the kick drum might just
arrive slightly quicker and more precisely, whether the ear/brain
detects it is another matter). I'm thinking it might not be so obvious
once we go through the science of it all...
Regards,
Arch
Shannon/Nyquist theory is by clear... To "perfectly"reconstruct a
signal of Frequency F you need a sampling frequency of 2F. The are of
course some subtle provisos to this theory. however there is definitely
no improved temporal resolution to be had - that is an audio myth, just
like the classically misleading (indeed poisonous) staircase diagram
that implies a need for lots more samples/bits... Coz you know 16/24 @
44.1 just ain't enough to capture the subtleties... :-->

The real question is is 20khz enough? I reckon that for 99.999% of
people who even care about this question, 15khz is plenty! :-)
--
Phil Leigh

You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1
DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's,
ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend
Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus
Interconnect cables
Stax4070+SRM7/II phones
Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything.
------------------------------------------------------------------------
Phil Leigh's Profile: http://forums.slimdevices.com/member.php?userid=85
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
vett93
2012-02-06 18:51:24 UTC
Permalink
Post by Phil Leigh
...
The real question is is 20khz enough? I reckon that for 99.999% of
people who even care about this question, 15khz is plenty! :-)
Do we still need supertweeters then? :-)
--
vett93

Main system:
Source: Transporter, modded by ModWright:
http://www.modwright.com/modifications/transporter-truth-mods.php
Preamp: Dude from Tube Research Labs:
http://www.tuberesearchlabs.com/products/dude.html
Amp: NP100 Platinum from AltaVista Audio:
http://www.altavistaaudio.com/np100.html
Speakers: Alto Utopia Be from Focal-JMLab:
http://www.focal.com/en/home-audio-loudspeakers/hifi-speakers/floorstanding-speakers/alto-utopia-be.php
------------------------------------------------------------------------
vett93's Profile: http://forums.slimdevices.com/member.php?userid=13301
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Phil Leigh
2012-02-06 18:55:02 UTC
Permalink
Post by vett93
Do we still need supertweeters then? :-)
Depends where they crossover :-)
--
Phil Leigh

You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1
DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's,
ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend
Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus
Interconnect cables
Stax4070+SRM7/II phones
Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything.
------------------------------------------------------------------------
Phil Leigh's Profile: http://forums.slimdevices.com/member.php?userid=85
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Wombat
2012-02-06 19:08:32 UTC
Permalink
Some interesting read about the temporal masking and its need or not for
higher sampling rate.
http://www.hydrogenaudio.org/forums/index.php?showtopic=73598&hl=impulse&st=0

You really have to read thru completely and try to understand as much
as possible. Me donŽt but find it interesting nonetheless :)
--
Wombat

Transporter (modded) -> RG142 -> Avantgarde Acoustic based 500VA
monoblocks -> Sommer SPK240 -> self-made speakers
------------------------------------------------------------------------
Wombat's Profile: http://forums.slimdevices.com/member.php?userid=4113
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Mnyb
2012-02-06 19:19:42 UTC
Permalink
You can of-course get rid of ADC artifacts lets say a not so god filter
implementation by recording in higher FS .

But down-sampling it to 44.1 should be transparent ?

The example is a bit theoretical ? normally a lot off post-processing
is used so working on 24/96 or 32-64 bit float in the software can't
hurt.

But I agree on the general principle (I could not do otherwise ,mr
nyqvist stands)

What about the bith dept ? 24 bit is a healthy marginal , was not some
older CD's more like 13bit and no dithering at all ? yuk.
--
Mnyb

--------------------------------------------------------------------
Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x
MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3
sub.
Bedroom/Office: Boom
Kitchen: Touch + powered Fostex PM0.4
Misc use: Radio (with battery)
iPad 64gB wifi +3g with iPengHD & SqueezePad
(in storage SB3, reciever ,controller )
------------------------------------------------------------------------
Mnyb's Profile: http://forums.slimdevices.com/member.php?userid=4143
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Wombat
2012-02-06 18:52:58 UTC
Permalink
Like Phil already pointed out there is lots of misinformation around
especialy by some audio hardware sellers that argue with impressive
pics.
Impulses in lowres suddenly look ugly and "look slow" what in reality
simply is the visualisation of a lowpass, nothing to wurry about.
When lame mp3 developement was in earlier state we once tested where to
set a lowpass and even the for sure way above average, younger (golden)
ears were not able to spot lowpassing in music at 18.5khz while they
could easily hear a 20kHz sinuid.
--
Wombat

Transporter (modded) -> RG142 -> Avantgarde Acoustic based 500VA
monoblocks -> Sommer SPK240 -> self-made speakers
------------------------------------------------------------------------
Wombat's Profile: http://forums.slimdevices.com/member.php?userid=4113
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
adamdea
2012-02-06 19:20:16 UTC
Permalink
This is a very good question. Hopefully Teddy Ray may be able to help
out with his very high powered friends.

I have been trying really hard to grasp the temporal resolution issue
and have found it really difficult. It seems to have been kicked off in
audiophile circles by the NOS dac bollocks and (separately and
indirectly) by an article by Craven for AES about the desirability of
apodising/ minimum phase filters [not quite the same thing I know].
These filters are somewhat in vogue although neither weiss nor
briscasti seem to use them.

Annoyingly I think that this is an incredibly difficult area and the
basic digital electronics text books i have read have very little on
it.

Just to get things started here are a couple of thoughts [APOLOGIES I
NOW REALISE I HAVE GONE OFF ON A BIT OF A RAMBLE BUT HERE GOES
ANYWAY}-

a. AFAIK there really isn't any evidence that we can hear above 20Khz
(make that 15 for most of us)

b. in order to be nyquist compliant before sampling at 44.1kHz (or
downsampling to 44.1), the signal has to be band limited to 22.05kHz or
lower.

c. there is an issue about whether the effect of this band limitation
could be audible.

d. we were talking here about the band limation prior to the ADC (or
resampling) NOT in your DAC.

e. but wait! the band limitation at the ADC has been created by a
number of processes starting with the mic and then (maybe the mixer etc
and the tape if its an old analog recording. So in order to analyse out
the effect of band limitation you have to look at this chain as a
whole

f. time domain and frequency domain analysis are basically two sides of
the same coin -if you limit the max frequency you must have an impact in
the time domain.

g. but this expression temporal resolution is a real bugger. AFAIK it
has no fixed meaning. It is bandied around by certain audiophiles as a
possible peg o which to hang their belief that "digital" especially
properly filtered digital doesn't sound good.

h if you don't believe me check out the kunchur/ JJ debate in which
Kunchur made an utter arse of himself, despite being a genuine
physicist, when he suggested that "temporal resolution" was limited to
the inter sample period.

i strictly speaking looking at the OP it seems to me that the answer is
that if there is no information in the original signal about 20Khz there
cannot be any time resolution issue with 20kHz band limitation. This is
because a proper filter should have no impact below the stop band. It
is therefore trivial that there could not be any information loss.

j. that answer might be a bit to smart arse though- the real question
is whether, assuming there was information above 20KhZ in the original
signal, the band limitation prior to sampling might still have some
detectable effect on the signal.

k Any band limitation must affect the signal in the time domain. In
particular the maximum frequency determines the maximum steepness of
the slope in the time domain. So the band limited signal has to be
spread out in the time domain if higher frequencies are filtered out.
BUT CAN ANYONE ACTUALLY HEAR THIS?
IF THEY CAN HEAR THE DIFFERENCE BETWEEN FIKLTERED AND UNFILTERED, the
IS ONE Sensible) FILTER BETTER THAN ANOTHER

l. this is where we get into pre ringng an post ringing. IN order to
make a filter which perfectly complies with nyquist it should in
principle be a linear phase filter. This will spread out the signal
equally in both directions (earlier and later) . But according to
auipophile magazines this apparently means pre ringing and this is
apparently bad.

m I would love to see
1) a demonstration of how a real world signal is affected by a linear
phase filter with a passband up to 20hZ or so and proper attenuation at
nyquist. Weirdly I have not actually seen one
2) some evidence that people can hear the difference between the
unfiltered and filtered signal and the difference between MP and LP
filters..
--
adamdea
------------------------------------------------------------------------
adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Wombat
2012-02-06 19:28:08 UTC
Permalink
Just create yourself such an impulse and apply a lowpass or resample it.
The post and preringing only happens around fs/2 so above 20kHz. I did
some and with a gentle filter kicking in around 20khz there is virtualy
no ringing left.
Of cause the picture of simply the impulse, not its spectrum is welcome
for marketing :)
--
Wombat

Transporter (modded) -> RG142 -> Avantgarde Acoustic based 500VA
monoblocks -> Sommer SPK240 -> self-made speakers
------------------------------------------------------------------------
Wombat's Profile: http://forums.slimdevices.com/member.php?userid=4113
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
adamdea
2012-02-06 19:34:25 UTC
Permalink
Post by Wombat
Just create yourself such an impulse and apply a lowpass or resample it.
The post and preringing only happens around fs/2 so above 20kHz. I did
some and with a gentle filter kicking in around 20khz there is virtualy
no ringing left.
Of cause the picture of simply the impulse, not its spectrum is welcome
for marketing :)
I agree that the impulse response is very misleading since it
represents the response of the filter to an impossible input.
BUt still the effect of pre-ringing is confusing. Although it
represents to effect of the filter cutting off frequencies above the
stop band AFAIK it supposedly spreads out energy throughout the
passband. This is where i get confused because it supposedly does so
even though the filter has a flat response in the passband- how can it
do that (PAGING TEDDY RAY)
--
adamdea
------------------------------------------------------------------------
adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Wombat
2012-02-06 19:44:09 UTC
Permalink
I agree that the impulse response is very misleading since it represents
the response of the filter to an impossible input.
BUt still the effect of pre-ringing is confusing. Although it
represents to effect of the filter cutting off frequencies above the
stop band AFAIK it supposedly spreads out energy throughout the
passband. This is where i get confused because it supposedly does so
even though the filter has a flat response in the passband- how can it
do that (PAGING TEDDY RAY)
I lately did some in a different forum, first some kind of Apodizing
against steep linear.
Loading Image.../
Second is a gentle linear:
Loading Image.../

You see it shouldnŽt touch the audible band at all.
--
Wombat

Transporter (modded) -> RG142 -> Avantgarde Acoustic based 500VA
monoblocks -> Sommer SPK240 -> self-made speakers
------------------------------------------------------------------------
Wombat's Profile: http://forums.slimdevices.com/member.php?userid=4113
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
adamdea
2012-02-06 20:23:37 UTC
Permalink
This is very intersting- what does the unfiltered signal look like?
--
adamdea
------------------------------------------------------------------------
adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Wombat
2012-02-06 20:30:56 UTC
Permalink
Original is 24/96 full scale impulse that looks exactly like the signal
below 20khz on these linears up to 48kHz.
--
Wombat

Transporter (modded) -> RG142 -> Avantgarde Acoustic based 500VA
monoblocks -> Sommer SPK240 -> self-made speakers
------------------------------------------------------------------------
Wombat's Profile: http://forums.slimdevices.com/member.php?userid=4113
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Phil Leigh
2012-02-06 19:30:48 UTC
Permalink
You also need to consider the sampling rate of the human ear and the
"inter-sample" interpolation (and all of the other heavyweight
processing) in the brain...
--
Phil Leigh

You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1
DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's,
ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend
Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus
Interconnect cables
Stax4070+SRM7/II phones
Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything.
------------------------------------------------------------------------
Phil Leigh's Profile: http://forums.slimdevices.com/member.php?userid=85
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
adamdea
2012-02-06 19:36:08 UTC
Permalink
Post by Phil Leigh
You also need to consider the sampling rate of the human ear and the
"inter-sample" interpolation (and all of the other heavyweight
processing) in the brain...
Absolutely, and i think that saying you can detect the time domain
effects is just sneaking through the back door the argument that you
can hear over 20KhZ
--
adamdea
------------------------------------------------------------------------
adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Phil Leigh
2012-02-06 19:43:59 UTC
Permalink
Post by adamdea
Absolutely, and i think that saying you can detect the time domain
effects is just sneaking through the back door the argument that you
can hear over 20KhZ
Agreed. Most of this is hogwash dreamt up by people who heard early
digital and felt it was bad (which it mostly was)... And then invented
a whole fake mythology about why it was bad and why it would never be
good.
Fools.
--
Phil Leigh

You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1
DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's,
ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend
Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus
Interconnect cables
Stax4070+SRM7/II phones
Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything.
------------------------------------------------------------------------
Phil Leigh's Profile: http://forums.slimdevices.com/member.php?userid=85
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
TheOctavist
2012-02-07 05:18:20 UTC
Permalink
The temporal resolution of 44.1 kHz audio is about

1/(44100*65536*2*pi) seconds, which is so far below any detectable
difference that I can not possibly imagine an ydifference.

I would love to stomp out this "time resolution" crap that we see all
the time in the high end, it's due to some people who don't understand
the concepts around sampling, how filtering works, etc
--
TheOctavist

Vortexbox>SBT(stock(TT failed dbt)>>Forssell MDAC-2>>>Klein and Hummell
0300D

Sota Sapphire/Lyra Kleos>>Bespoke Valve Phono Stage>>Mastersound Due
Venti>>Link Audio K100
------------------------------------------------------------------------
TheOctavist's Profile: http://forums.slimdevices.com/member.php?userid=52700
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Archimago
2012-02-07 06:00:32 UTC
Permalink
Awesome discussion guys! I must admit a lot of the science is over my
head but great to read so many knowledgeable posts; clearly discussion
an order of magnitude beyond much of the bickering over the last couple
weeks.

At 40 years old now, my frequency threshold is around 17kHz. I've
ABX'ed 24/96 songs downsampled to 16/44 through pretty reasonable
hardware attached to my main computer (E-Mu0404USB --> Music Hall
Ph25.2 headphone amp --> Sennheiser HD800) unsuccessfully. The more I
test this out for myself, the less I'm tempted to buy any more 'hi-res'
music.

I really do hope the loudness war ends at some point to elevate the
mastering job on more modern music as that is what seems to be the real
'quality limiting' piece these days. I suspect that much of the raving
around hi-res music whether DVD-A or SACD has been the fact that they
were often mastered at a higher level if the science is pretty clear
that ears/brains are unlikely to experience an improvement just from
the specs...
--
Archimago
------------------------------------------------------------------------
Archimago's Profile: http://forums.slimdevices.com/member.php?userid=2207
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Mnyb
2012-02-07 06:14:06 UTC
Permalink
If you try hf test tones at home make sure the DAC works properly .

I tried with my m-audio computer speakers, but my old soundblaster
souncards is not up to it or I have severe audio subsystem problems on
the desktop or both.

I get aliasing or similar artefacts you can definitely hear something
when playing tones close to 20k :)

I'll bet some of the cargo cult engineered nos dacs or similar also
would fail in this respects so using properly designed equipment is
paramount as usual.
--
Mnyb

--------------------------------------------------------------------
Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x
MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3
sub.
Bedroom/Office: Boom
Kitchen: Touch + powered Fostex PM0.4
Misc use: Radio (with battery)
iPad 64gB wifi +3g with iPengHD & SqueezePad
(in storage SB3, reciever ,controller )
------------------------------------------------------------------------
Mnyb's Profile: http://forums.slimdevices.com/member.php?userid=4143
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
finnbrodersen
2012-02-11 13:31:56 UTC
Permalink
my old ears stop at about 12-13kHz
--
finnbrodersen

Some version of SBServer running on a HP EX490 home server
SBReceiver --> NAD C162+C272 --> DALI IKON 6 (let's call it MidFi)
------------------------------------------------------------------------
finnbrodersen's Profile: http://forums.slimdevices.com/member.php?userid=17360
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
evdplancke
2012-02-06 20:41:13 UTC
Permalink
Nyquist theorema tells there is no benefits but is based on the
following assumptions:
1) sampled signal is made of perfect dirac pulses... practically they
will look like square pulses resulting in a quite bad low pass sin x/x
filtering that zeroes at 1/T where T is the width of the pulse
(staircase sampled signal being the worst with 1/T equals to sample
freq)
2) sampled pulse has a constant time period... but in reality time
period is varying due to jitter
3) signal reconstruction is using perfect brickwall filters... but in
practice it is very difficult to have steep filter slope without
artifacts

In conclusion, there might be a (big?) gap between theory and practice.

According to the 3 assumptions above, here are the (audible?) benefits
of 96khz:
1) higher sample freq is reducing the low pass effect of square pulses,
rejecting the resulting zeroes to higher freq
2) doubling the number of pulses adds redundancy to the signal that
increases immunity to jitter
3) less steep brickwall filters are needed for signal reconstruction,
reducing artifacts and improving linearity

So hi res might technically improve the accuracy of reconstructed
signal even if baseband signal does not contain any freq above 20 khz.

The question remains however: are those benefits audible or not? I'd
say it makes it at least more robust against less than optimal receiver
implementation.
--
evdplancke


------------------------------------------------------------------------
evdplancke's Profile: http://forums.slimdevices.com/member.php?userid=43147
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Phil Leigh
2012-02-06 20:48:09 UTC
Permalink
Post by evdplancke
Nyquist theorema tells there is no benefits but is based on the
1) sampled signal is made of perfect dirac pulses... practically they
will look like square pulses resulting in a quite bad low pass sin x/x
filtering that zeroes at 1/T where T is the width of the pulse
(staircase sampled signal being the worst with 1/T equals to sample
freq)
2) sampled pulse has a constant time period... but in reality time
period is varying due to jitter
3) signal reconstruction is using perfect brickwall filters... but in
practice it is very difficult to have steep filter slope without
artifacts
In conclusion, there might be a (big?) gap between theory and
practice.
According to the 3 assumptions above, here are the (audible?) benefits
1) higher sample freq is reducing the low pass effect of square pulses,
rejecting the resulting zeroes to higher freq
2) doubling the number of pulses adds redundancy to the signal that
increases immunity to jitter
3) less steep brickwall filters are needed for signal reconstruction,
reducing artifacts and improving linearity
So hi res might technically improve the accuracy of reconstructed
signal even if baseband signal does not contain any freq above 20 khz.
The question remains however: are those benefits audible or not? I'd
say it makes it at least more robust against less than optimal receiver
implementation.
There's still no irrefutable evidence that anyone can hear the
difference - you'd think by now the Net would have at least a few
conclusive studies....?
--
Phil Leigh

You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1
DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's,
ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend
Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus
Interconnect cables
Stax4070+SRM7/II phones
Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything.
------------------------------------------------------------------------
Phil Leigh's Profile: http://forums.slimdevices.com/member.php?userid=85
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
adamdea
2012-02-06 21:40:53 UTC
Permalink
Post by evdplancke
Nyquist theorema tells there is no benefits but is based on the
1) sampled signal is made of perfect dirac pulses... practically they
will look like square pulses resulting in a quite bad low pass sin x/x
filtering that zeroes at 1/T where T is the width of the pulse
(staircase sampled signal being the worst with 1/T equals to sample
freq)
2) sampled pulse has a constant time period... but in reality time
period is varying due to jitter
3) signal reconstruction is using perfect brickwall filters... but in
practice it is very difficult to have steep filter slope without
artifacts
In conclusion, there might be a (big?) gap between theory and
practice.
According to the 3 assumptions above, here are the (audible?) benefits
1) higher sample freq is reducing the low pass effect of square pulses,
rejecting the resulting zeroes to higher freq
2) doubling the number of pulses adds redundancy to the signal that
increases immunity to jitter
3) less steep brickwall filters are needed for signal reconstruction,
reducing artifacts and improving linearity
So hi res might technically improve the accuracy of reconstructed
signal even if baseband signal does not contain any freq above 20 khz.
The question remains however: are those benefits audible or not? I'd
say it makes it at least more robust against less than optimal receiver
implementation.
I am not really sure what you mean by 1)except as a corollary of 3).
and I'm not sure about 2) either -surely a higher sampling rate would
require more accurate timing in the ADC? not that I think this is an
issue.

The real point is that doubling fs halves the quantisation noise in the
audible spectrum.This the equivalent of adding 1 bit albeit very
inefficiently since you have doubled the data rate whilst increasing
the amount of information by the same amount you could have achieved by
adding one bit which would have increased the data rate by 1/16.It does
give you some room to play with for noise shaping though.

I think 3) is the real time domain issue. But it's worth pointing out
that the fundamental issue is that increasing fs enables you to have
less steep filter in the anti alias stage of ADC. But it all begs the
question - given the ability to create very steep digital filters with
negligible passband ripple or phase issues, what are the artefacts?

I note however that daniel weiss indicated in an interview that he
thought 16/88 would be more beneficial as a consumer format than
24/48. I don't think he spelt this out but I assume it was because he
thought there was chance that there was something in the time domain
issues.

Even then though it's worth stressing that there should not be any time
domain issue if the pre-filtering pre ADC signal has no energy above
20kHz - there will not be any time smear then![I Hold my breath and
wait to be shown up as an idiot]

Equally If there is no signal above 20KHz in the 24/96 file then either
there was none in the pre recording sound [no time smear with sensible
filtering] or there was but it has been filtered out anyway ie the time
smear will have occured. . IN that case I can't see how there could be
any worthwhile improvement in a 96kHz file over a 44.1 kHz.
--
adamdea
------------------------------------------------------------------------
adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
evdplancke
2012-02-06 23:24:34 UTC
Permalink
Post by adamdea
I am not really sure what you mean by 1)except as a corollary of 3).
1) is different from 3) because the low pass effect will affect the
baseband signal: a convolution of a square pulse with perfect dirac
pulse train in the time domain corresponds to multiplying the signal in
the frequency domain by a very poor sinx/x lowpass filter. This has
nothing to do with the reconstruction filter used for 3): this is a
lowpass distortion BEFORE reconstruction, that is definitely altering
the baseband signal in an extent inversely proportional to the square
pulse width.
--
evdplancke


------------------------------------------------------------------------
evdplancke's Profile: http://forums.slimdevices.com/member.php?userid=43147
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
regalma1
2012-02-07 00:18:08 UTC
Permalink
There are a number of rules in the audio world that haven't held up. For
long time it was considered that we could only hear differences of at
least 1 dB. That has been demonstrated to not be correct. I remember
reading about a double blind test in Stereo Review that went bad
because there was a .1 dB diffenence in levels. The listeners could
pick it up. Bell Labs was able to show that filtering out everything
above 30 KHz was detectable by a person who was tone deaf above 10 KHz.
Jim Smith in his book "Get Better Sound" has a great story about
discovering that he could pick out sounds that were below the noise
floor in analog recordings but not in a digital recording of the same.
The human ear is pretty amazing.

The naysayers about cables always claim that only resistance and
reactance count. We in the wireless industry are being increasingly
plagued by a distortion called PIM, which occurs in cables and
connectors. It is every bit as possible in the audio range as the
microwave. And no one except maybe a specially built research lab can
measure PIM in the audio frequency range. We only recently have
developed the ability to measure it easily in the microwave range,
which is orders of magnitude easier.

If people are consistently hearing differences even though the theory
or instruments say they shouldn't we should assume it is always a case
of self delusion. We knowledge is dwarfed by our ignorance.
--
regalma1
------------------------------------------------------------------------
regalma1's Profile: http://forums.slimdevices.com/member.php?userid=6658
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Wombat
2012-02-07 01:02:21 UTC
Permalink
Post by regalma1
There are a number of rules in the audio world that haven't held up. For
long time it was considered that we could only hear differences of at
least 1 dB. That has been demonstrated to not be correct. I remember
reading about a double blind test in Stereo Review that went bad
because there was a .1 dB diffenence in levels. The listeners could
pick it up. Bell Labs was able to show that filtering out everything
above 30 KHz was detectable by a person who was tone deaf above 10 KHz.
Jim Smith in his book "Get Better Sound" has a great story about
discovering that he could pick out sounds that were below the noise
floor in analog recordings but not in a digital recording of the same.
Please link to some serious study that claims only steps of 1db are
audible. Of cause it is much smaller steps and when you are at it the
study that shows 0.1dB. Same goes for the signal filtered at 30Khz.
--
Wombat

Transporter (modded) -> RG142 -> Avantgarde Acoustic based 500VA
monoblocks -> Sommer SPK240 -> self-made speakers
------------------------------------------------------------------------
Wombat's Profile: http://forums.slimdevices.com/member.php?userid=4113
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
darrell
2012-02-07 01:05:41 UTC
Permalink
Post by regalma1
There are a number of rules in the audio world that haven't held up. For
long time it was considered that we could only hear differences of at
least 1 dB. That has been demonstrated to not be correct. I remember
reading about a double blind test in Stereo Review that went bad
because there was a .1 dB diffenence in levels. The listeners could
pick it up. Bell Labs was able to show that filtering out everything
above 30 KHz was detectable by a person who was tone deaf above 10 KHz.
Jim Smith in his book "Get Better Sound" has a great story about
discovering that he could pick out sounds that were below the noise
floor in analog recordings but not in a digital recording of the same.
The human ear is pretty amazing.
The naysayers about cables always claim that only resistance and
reactance count. We in the wireless industry are being increasingly
plagued by a distortion called PIM, which occurs in cables and
connectors. It is every bit as possible in the audio range as the
microwave. And no one except maybe a specially built research lab can
measure PIM in the audio frequency range. We only recently have
developed the ability to measure it easily in the microwave range,
which is orders of magnitude easier.
If people are consistently hearing differences even though the theory
or instruments say they shouldn't we should assume it is always a case
of self delusion. We knowledge is dwarfed by our ignorance.
There is an easy answer to all this: a statistically significant double
blind test. If such a test proves to an acceptable degree of certainty
*that* a difference exists, then we can move on to investigating *why*
it exists. But not until then, because there are an infinite number of
things we could "assume", and we don't have the time to investigate
then all.
--
darrell
------------------------------------------------------------------------
darrell's Profile: http://forums.slimdevices.com/member.php?userid=13460
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Wombat
2012-02-07 01:22:55 UTC
Permalink
Since here are so many members around that really believe they are above
golden eared there is one big chance:
IŽd like the people from here bringing samples and honest testing
showing to us all what the higher bandwith brings.
Try to resample a version of your beloved showcase and offer your
findings here.
Do some foobar abx with a headset if you like at home and post the logs
with all honesty here. Please no liquid soundstage bullshit, just plain
abx. It is simple.
Afterwards we can collect the positive results and dig deeper why you
were positive. We can help each other with preparing the files.
If all goes positive we may ask on hydrogenaudio for further backup. I
think this can be some fun.
I did a kind of this lately :)
--
Wombat

Transporter (modded) -> RG142 -> Avantgarde Acoustic based 500VA
monoblocks -> Sommer SPK240 -> self-made speakers
------------------------------------------------------------------------
Wombat's Profile: http://forums.slimdevices.com/member.php?userid=4113
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
adamdea
2012-02-07 01:39:05 UTC
Permalink
Post by darrell
There is an easy answer to all this: a statistically significant double
blind test. If such a test proves to an acceptable degree of certainty
*that* a difference exists, then we can move on to investigating *why*
it exists. But not until then, because there are an infinite number of
things we could "assume", and we don't have the time to investigate
then all.
AFAIK contrary to what regalma1 says there have been studies which have
shown that people can tell properly low pass filtered material from the
original with 40 kHz or whatever bandwidth.

Plus we have to wonder why adults are not bothered by anti teenager
mosquito devices, ultrasonic machinery or indeed noise shaping,
--
adamdea
------------------------------------------------------------------------
adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
TheOctavist
2012-02-07 01:58:27 UTC
Permalink
ive asked all my elite audio friends, Adam.

more to follow.
--
TheOctavist

Vortexbox>SBT(stock(TT failed dbt)>>Forssell MDAC-2>>>Klein and Hummell
0300D

Sota Sapphire/Lyra Kleos>>Bespoke Valve Phono Stage>>Mastersound Due
Venti>>Link Audio K100
------------------------------------------------------------------------
TheOctavist's Profile: http://forums.slimdevices.com/member.php?userid=52700
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
adamdea
2012-02-07 09:15:01 UTC
Permalink
Post by TheOctavist
ive asked all my elite audio friends, Adam.
more to follow.
Fantastic TR
I think there are 2 related issues
-whether >44.1 has any advantages for playback
- whether there are advantages in any particular type of filter for
reconstruction downsampling or bandlimiting (assuming we already have a
standard linear phase half band filter)
--
adamdea
------------------------------------------------------------------------
adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
adamdea
2012-02-10 11:35:46 UTC
Permalink
Post by TheOctavist
ive asked all my elite audio friends, Adam.
more to follow.
Hi TR, did you get any further reply from your pals?
My it has gone quiet all of a sudden on this forum.
--
adamdea
------------------------------------------------------------------------
adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
TheOctavist
2012-02-11 12:36:20 UTC
Permalink
Post by adamdea
Hi TR, did you get any further reply from your pals?
My it has gone quiet all of a sudden on this forum.
Not yet..but I will soon! if not, ill phone em. :)


btw, Adam, if you have any specific in depth questions...let me know,
ill send em up the chain...and for that matter, the legendary engineer
Tony Faulkner helps me a lot too, so if you have any recording
engineering questions...
--
TheOctavist

Vortexbox>SBT(stock(TT failed dbt)>>Forssell MDAC-2>>>Klein and Hummell
0300D

Sota Sapphire/Lyra Kleos>>Bespoke Valve Phono Stage>>Mastersound Due
Venti>>Link Audio K100
------------------------------------------------------------------------
TheOctavist's Profile: http://forums.slimdevices.com/member.php?userid=52700
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
rgro
2012-02-07 02:49:48 UTC
Permalink
Post by adamdea
Plus we have to wonder why adults are not bothered by anti teenager
mosquito devices, ultrasonic machinery or indeed noise shaping,
I have to disagree with at least the first part of this statement.
About 4 years ago, I was curious about this mosquito device (in my
mid-50's, then)---had just read something about it----and found a
number of audio clips of them on websites.

So, I played a few and it bothered me plenty (and I took a blind test
with my teenaged daughters administering it and passed with 100%
accuracy)! Don't know whether or not I could hear it now, but I suspect
I'm not the only "adult" that could hear it. My wife could not, though,
and was absolutely convinced---much to our great hilarity---that we were
playing mean tricks on her.
--
rgro

Rg

System information
------------------------
Main: PS Audio Quintet > Vortexbox > Teddy Pardo PS, Touch (wired) >
Toslink > Rega DAC > LFD LE IV Signature amp > VA Mozart Grands > REL
Acoustics R305.

Home Theatre: Duet/SBR (Wired) > Pioneer VSX 919 > Energy Take 5
Classic 5.1.

SBS 7.7.1 r33751 on a Vortexbox Appliance, V 2.0, Touch: FW 7.7.1
r9558. Duet: FW 7.7.1 r9557.
------------------------------------------------------------------------
rgro's Profile: http://forums.slimdevices.com/member.php?userid=34348
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Wombat
2012-02-07 03:04:46 UTC
Permalink
AFAIK the mentioned mosquito handy tone is at around 17kHz. Some years
ago when it was trendy i tried these tones and heard them clearly.
--
Wombat

Transporter (modded) -> RG142 -> Avantgarde Acoustic based 500VA
monoblocks -> Sommer SPK240 -> self-made speakers
------------------------------------------------------------------------
Wombat's Profile: http://forums.slimdevices.com/member.php?userid=4113
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
rgro
2012-02-07 03:13:17 UTC
Permalink
Post by Wombat
AFAIK the mentioned mosquito handy tone is at around 17kHz. Some years
ago when it was trendy i tried these tones and heard them clearly.
According to Wikipedia it's 17.4kHz. Glad to know there's more than
one old fart whose ears still work o.k.!
--
rgro

Rg

System information
------------------------
Main: PS Audio Quintet > Vortexbox > Teddy Pardo PS, Touch (wired) >
Toslink > Rega DAC > LFD LE IV Signature amp > VA Mozart Grands > REL
Acoustics R305.

Home Theatre: Duet/SBR (Wired) > Pioneer VSX 919 > Energy Take 5
Classic 5.1.

SBS 7.7.1 r33751 on a Vortexbox Appliance, V 2.0, Touch: FW 7.7.1
r9558. Duet: FW 7.7.1 r9557.
------------------------------------------------------------------------
rgro's Profile: http://forums.slimdevices.com/member.php?userid=34348
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Wombat
2012-02-07 03:25:17 UTC
Permalink
According to Wikipedia it's 17.4kHz. Glad to know there's more than one
old fart whose ears still work o.k.!
Heh! I bet many still can hear this. Nonetheless i have to admit the
hearing slowly degrades no matter how much i try to guard my ears. Some
things simply donŽt jump on me soundwise like before. I was used to hear
several hard to hear glitches in mp3 encoding with ease. The same kind
of glitches become harder for me. Still i can hear up to ~18kHz with my
early 40s.
I sometimes wonder how some of the reviewers that often must be in the
60s claim all kinds of things they still hear.
--
Wombat

Transporter (modded) -> RG142 -> Avantgarde Acoustic based 500VA
monoblocks -> Sommer SPK240 -> self-made speakers
------------------------------------------------------------------------
Wombat's Profile: http://forums.slimdevices.com/member.php?userid=4113
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Mnyb
2012-02-07 05:40:11 UTC
Permalink
Are we mixing two things in this tread recording and playback ?

Are they not all kinds of very good reasons to record at higher rates
24/96 produce the thing and then properly normalise downsample and
dither and have a well produced 16/44.1 CD .
And this sound just fine , it is very likely that most of us would fail
to abx this.

Is this not in fact the typical mo of modern recording ? ( with the
destructive loudness war phase removed )

Dow sampling well done modern hirez production is one thing to abx .
( not one off hdcraps 45y old analog rock classics I bet here 14bits
are ok ).

I can donate some itrax tunes ( aix ) for the purpose or a chesky .

But recording all tracks at 16/44.1 and then produce the whole shebang
at this rate ?
And do the same process at higher rate in parallell ? This would be
interesting to abx.

The pro's that are in this tread, are there any ADC's you would
consider transparent especially one that also can do 16/44.1 as oposed
to the modern 24/48-192k setting and have > 1 of them ?

Wombat or phil may be able to find studio time ? Afaik your studio
software would remeber what you did when producing so repeating these
steps would be feasible.

What would a modern DAW do whith 16/44.1 original data would it
upsample and turn it to 32 bit floating piont data so that all plugins
eq and stuff would work transparently ? Can this be defeated ?

Anyone any good at singing or playing ?
--
Mnyb

--------------------------------------------------------------------
Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x
MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3
sub.
Bedroom/Office: Boom
Kitchen: Touch + powered Fostex PM0.4
Misc use: Radio (with battery)
iPad 64gB wifi +3g with iPengHD & SqueezePad
(in storage SB3, reciever ,controller )
------------------------------------------------------------------------
Mnyb's Profile: http://forums.slimdevices.com/member.php?userid=4143
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
TheOctavist
2012-02-07 06:05:54 UTC
Permalink
Post by Mnyb
Anyone any good at singing or playing ?
im an engineer, and have nice recording gear. I use MYTEK AD/DA
transparent. superb.

what do you need?
--
TheOctavist

Vortexbox>SBT(stock(TT failed dbt)>>Forssell MDAC-2>>>Klein and Hummell
0300D

Sota Sapphire/Lyra Kleos>>Bespoke Valve Phono Stage>>Mastersound Due
Venti>>Link Audio K100
------------------------------------------------------------------------
TheOctavist's Profile: http://forums.slimdevices.com/member.php?userid=52700
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
TheOctavist
2012-02-07 06:08:00 UTC
Permalink
Post by Mnyb
What would a modern DAW do whith 16/44.1 original data would it
upsample and turn it to 32 bit floating piont data so that all plugins
eq and stuff would work transparently ? Can this be defeated ?
Anyone any good at singing or playing ?
yes, most all use 32 bit FP, other than REAPER(which I think uses 64)

it can be disabled on some, yes. like my sequoia workstation(which also
allows me to hear the two rates simultaneously..
--
TheOctavist

Vortexbox>SBT(stock(TT failed dbt)>>Forssell MDAC-2>>>Klein and Hummell
0300D

Sota Sapphire/Lyra Kleos>>Bespoke Valve Phono Stage>>Mastersound Due
Venti>>Link Audio K100
------------------------------------------------------------------------
TheOctavist's Profile: http://forums.slimdevices.com/member.php?userid=52700
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Wombat
2012-02-07 16:43:16 UTC
Permalink
Post by Mnyb
Are we mixing two things in this tread recording and playback ?
Are they not all kinds of very good reasons to record at higher rates
24/96 produce the thing and then properly normalise downsample and
dither and have a well produced 16/44.1 CD .
And this sound just fine , it is very likely that most of us would fail
to abx this.
Is this not in fact the typical mo of modern recording ? ( with the
destructive loudness war phase removed )
Dow sampling well done modern hirez production is one thing to abx .
( not one off hdcraps 45y old analog rock classics I bet here 14bits
are ok ).
I can donate some itrax tunes ( aix ) for the purpose or a chesky .
But recording all tracks at 16/44.1 and then produce the whole shebang
at this rate ?
And do the same process at higher rate in parallell ? This would be
interesting to abx.
The pro's that are in this tread, are there any ADC's you would
consider transparent especially one that also can do 16/44.1 as oposed
to the modern 24/48-192k setting and have > 1 of them ?
Wombat or phil may be able to find studio time ? Afaik your studio
software would remeber what you did when producing so repeating these
steps would be feasible.
What would a modern DAW do whith 16/44.1 original data would it
upsample and turn it to 32 bit floating piont data so that all plugins
eq and stuff would work transparently ? Can this be defeated ?
Anyone any good at singing or playing ?
I am no pro but recording, editing, applying effects of cause should be
done at the highest rate it is comfortable to work with. Why not if the
power is simply there with todays hardware.
I did read about several plugins for VST hosts that indeed work at
higher sampling and bitrate internaly.
For the final distribution to us listeners i still think anything
higher as 16/44.1 will hardly give us any benefit. Of cause when this
last step is done correctly.
--
Wombat

Transporter (modded) -> RG142 -> Avantgarde Acoustic based 500VA
monoblocks -> Sommer SPK240 -> self-made speakers
------------------------------------------------------------------------
Wombat's Profile: http://forums.slimdevices.com/member.php?userid=4113
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Jeff Flowerday
2012-02-08 13:47:55 UTC
Permalink
So here's a question somewhat related

My DAC has multiple filter settings, I'm interested in why they chose
any 2 parituclar filter combination to apply for low and high res.

More specifically the last 2 settings:

Setting 5, Minimum Phase Apodizing used for low res, anything above
24/48 uses Linear Phase Apodizing. Or it could be toggled to do the
exact opposite with setting 4.

http://www.stevehoffman.tv/forums/archive/index.php/t-239298.html

http://hifiduino.blogspot.com/2009/05/wm8741-digital-filters.html
--
Jeff Flowerday
------------------------------------------------------------------------
Jeff Flowerday's Profile: http://forums.slimdevices.com/member.php?userid=15883
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Wombat
2012-02-08 16:16:03 UTC
Permalink
We had this in another thread already. At Steve Hoffmans forums you can
read a lot fuzzy statements, so donŽt give that to much weight.
The other thread shows exctly the misleading pics i mentioned before.
Make these pics spectral and youŽll see all this post and pre stuff
happens well above 20kHz if done correctly. Even the aliasing back can
be above 20kHz, no problem. No one can really hear if the noise above
20Khz is music or aliasiad garbage. You may try to create a file with
no content below 20kHz and abx it, good luck!! :)
Also these apodizing and minimum or intermediate phase as far as i
tested them clearly alter frequencies below 20kHz and may be audible
just because of that. You wonŽt see that on that funny impulse pic.
Also imagine the studio engineer already did the downsampling with some
minimun filter and now you apply one again? In theory you must fiddle
with a setting for every single recording.
One other thing is that if all the junk above 20kHz changes the way
your audio equipment sounds it is time to wonder.
--
Wombat

Transporter (modded) -> RG142 -> Avantgarde Acoustic based 500VA
monoblocks -> Sommer SPK240 -> self-made speakers
------------------------------------------------------------------------
Wombat's Profile: http://forums.slimdevices.com/member.php?userid=4113
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Mnyb
2012-02-08 16:54:28 UTC
Permalink
Post by Wombat
I am no pro but recording, editing, applying effects of cause should be
done at the highest rate it is comfortable to work with. Why not if the
power is simply there with todays hardware.
I did read about several plugins for VST hosts that indeed work at
higher sampling and bitrate internaly.
For the final distribution to us listeners i still think anything
higher as 16/44.1 will hardly give us any benefit. Of cause when this
last step is done correctly.
That was my piont, it has merits to produce and record in hirez,
however the delivery format to consumer is another question .

In every tread we discuss this recording producing and playback is
mixed up, they deserve separat treatment .

I do believe in 24/96 or 24/48 for another reason , eq and processing
does not stop in the recording studio anymore , I have at least 6
digital xover pionts DRC Eq and spatial processing, digital volume
controll going on in my home theater .
To Meridians credit it can be said that this seems to work very well
with any source material, the benefit may be more theoretical and
possibly measurable , but audiability may be another question.
--
Mnyb

--------------------------------------------------------------------
Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x
MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3
sub.
Bedroom/Office: Boom
Kitchen: Touch + powered Fostex PM0.4
Misc use: Radio (with battery)
iPad 64gB wifi +3g with iPengHD & SqueezePad
(in storage SB3, reciever ,controller )
------------------------------------------------------------------------
Mnyb's Profile: http://forums.slimdevices.com/member.php?userid=4143
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
adamdea
2012-02-07 01:10:00 UTC
Permalink
Post by evdplancke
1) is different from 3) because the low pass effect will affect the
baseband signal: a convolution of a square pulse with perfect dirac
pulse train in the time domain corresponds to multiplying the signal in
the frequency domain by a very poor sinx/x lowpass filter. This has
nothing to do with the reconstruction filter used for 3): this is a
lowpass distortion BEFORE reconstruction, that is definitely altering
the baseband signal in an extent inversely proportional to the square
pulse width.
Are you referring here to aperture effect? I understand that this can
be a problem, but can be compensated by equalisation or resampling to
reduce the aperture ratio. But this is not resampling in the sense of
increasing the sample rate. Wouldn't having a higher sample rates mean
that your pulse train would need to be more perfect - a given pulse
duration would now represent a higher proportion of the sample period
surely(?) and would therefore reach zero order hold for a shorter pulse
width .
--
adamdea
------------------------------------------------------------------------
adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
pski
2012-02-07 01:17:56 UTC
Permalink
Theoretical is

If you say you hear this: NUTS !
--
pski

real stereo doesn't just wake the neighbors, it -enrages- them.. It is
truly the Golden Age of Wireless
------------------------------------------------------------------------
pski's Profile: http://forums.slimdevices.com/member.php?userid=15574
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
TheOctavist
2012-02-13 21:20:34 UTC
Permalink
From Mr. Putzeys.
Nope. If there is absolutely no content above a certain f, any sampling
rate of 2*f or over will reconstruct the exact same signal. Sampling
theory proves that a correctly executed (=filtered before and after)
sampling conversion system is indistinguishable from a lowpass filter.
And a lowpass filter is indistinguishable from nothing at all if the
signal has nothing for the lowpass filter to remove. The latter assumes
linear phase lowpass filters of course, but those are standard in AD/DA
these days.

Needless to say, when there is content above 20kHz the question gets
hairier.

a. In formal (double blind) trials, no results have yet been reported
of listeners able to distinguish between a "high sampling rate" audio
signal and same with a lowpass filter inserted. Note that the ear is
not a linear device. That is to say, you can't draw conclusions
regarding audibility of HF spectral content based on audibility of
single tone stimuli. For instance, quite a few people whose hearing
ostensibly stops at 10k are able to detect the insertion of a 15kHz
lowpass filter. But it does appear that 20kHz is roughly where it
ends.

b, c. I do mean roughly. If you make the filter very sharp and let it
cut off right at 20kHz it can become ridiculously obvious. A 4kHz
transition band is more or less what you need to render it inaudible.
Trouble is: 20kHz+4kHz>44.1kHz/2... Actually most filters do have a
4.1kHz transition band on account of cost saving (look up "halfband
filter). What this means is that most converters do not satisfy the
Nyquist criterium. This is not very audible, but still the world would
have been a different place had we standardized on 48kHz with a
non-halfband filter with a 4kHz transition band. The whole Hirez thing
would have gone nowhere.

g. Exactly. Count on it that most lay people think that sampling means
that you can't know the moment something happens (e.g. zero crossings)
with better precision that one sampling period. That's obvious
bollocks. It is precisely the LPF's task to insure that such
information is coded with essentially infinite precision (same limit as
an ordinary noise limited low-pass filtered channel). There is no
deficiency in the ability of digital audio to encode timing. The only
"resolution" that high sampling rates add is the ability to distinguish
two closely spaced events (as in, one sampling period apart). But that
argument is better carried through in the frequency domain.

I think that for your own sanity, whenever people are debating what is
essentially a specialised subject using vague terminology, you should
simply say "to each his own" and wander off. People who "debate"
sampling theory are not that much different from those who "debate"
evolution theory. It too is a remarkably precise and powerful theory
which is only being debated by lay people because they have an
emotional stake in its consequences.

Regarding NOS: such converters have a frequency response of
sin(pi*f/fs)/(pi*f/fs). This causes its characteristic sound. A friend
of mine who is a mastering engineer once came raving about some box by
Altmann. So I told him to upsample his audio 2x, then mix it with
itself shifted 1 sample, take it back down 2x and listen what it does
on a real DAC. The result was as predicted. This signal through his
normal DAC sounded exactly like the original signal through the
Altmann. Theory wins, again.

Regarding apodizing: I saw Peter's presentation of that. He does with
plastic slides stuff that I've yet to see a powerpoint rodeo do.
Essentially he takes two premises from the opinion circuit without
passing judgment but merely looks where they lead, to wit: "what if the
transition band determines audibility" and "what if pre-ringing is
audible, and inband phase shift is audible too". Based on these two he
then designs filters that are flat in magnitude and linear in phase up
to 20kHz and then roll off gently to hit zero at fs/2. Now, if you look
at it closely this means that his design method as applied to a 44.1kHz
sampling rate will yield an ordinary sharp, linear-phase filter. You
need a high sampling rate for apodizing filters to make sense. At 96kHz
the filter starts rolling off gently towards cut-off at 48kHz. At that
point the question becomes: OK so how does this compare with a sharp
filter cutting off at 40kHz? Ri-ight. So after finding it hard to
obtain solid double-blind data for 20kHz band-limited audio we're now
about to embark on a quest for the ideal filter at 40kHz. You can
imagine why it's not catching on.
--
TheOctavist

Vortexbox>SBT(stock)>>Forssell MDAC-2>>>Klein and Hummell 0300D

Sota Sapphire/Lyra Kleos>>Bespoke Valve Phono Stage>>Mastersound Due
Venti>>Link Audio K100
------------------------------------------------------------------------
TheOctavist's Profile: http://forums.slimdevices.com/member.php?userid=52700
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
adamdea
2012-02-14 16:52:26 UTC
Permalink
wow thanks very much- this is really interesting.
Please say thank you from me. I shall have a good read and properly
digest it before asked any dumb questions back.
--
adamdea
------------------------------------------------------------------------
adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Archimago
2012-02-16 02:41:38 UTC
Permalink
Wow, great post Octavist!
--
Archimago
------------------------------------------------------------------------
Archimago's Profile: http://forums.slimdevices.com/member.php?userid=2207
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
evdplancke
2012-02-16 16:01:13 UTC
Permalink
Thanks for your very interesting post. So we can conclude from there:
1) there is a benefit of using 48khz instead of 44,1khz because of the
reconstruction filter
2) fanatics of NOS DAC will benefit from hires even up to 384khz
because the higher the fs, the flatter the sin x/x lowpass in audible
frequency band...

Then what about a NOS DAC without output filtering (using ear natural
lowpass filtering): this would get rid of pre-ringing filtering
artefacts. Combined with ultra hires, wouldn't it be close to the
perfect sound?
--
evdplancke


------------------------------------------------------------------------
evdplancke's Profile: http://forums.slimdevices.com/member.php?userid=43147
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Wombat
2012-02-16 16:23:04 UTC
Permalink
Post by evdplancke
1) there is a benefit of using 48khz instead of 44,1khz because of the
reconstruction filter
2) fanatics of NOS DAC will benefit from hires even up to 384khz
because the higher the fs, the flatter the sin x/x lowpass in audible
frequency band...
Then what about a NOS DAC without output filtering (using ear natural
lowpass filtering): this would get rid of pre-ringing filtering
artefacts. Combined with ultra hires, wouldn't it be close to the
perfect sound?
To 1) there is still this theoretical filterproblem that even in
real-world isnŽt. It is only a problem for many because it is there
theoretical. Can you hear the different filters acting around 20kHz?

To 2) i never listened to a NOS DAC and many seem to love it while
others find even the best NOS DAC sounds horrible. The music it spits
out without sinc filter is indeed horrible to measure and for sure will
benefit of higher sampling frequencies most. Now this is a case to
wonder, even when the output is technicaly speaking pretty less precise
it still may be sounding fantastic to some.
--
Wombat

Transporter (modded) -> RG142 -> Avantgarde Acoustic based 500VA
monoblocks -> Sommer SPK240 -> self-made speakers
------------------------------------------------------------------------
Wombat's Profile: http://forums.slimdevices.com/member.php?userid=4113
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Phil Leigh
2012-02-16 17:54:23 UTC
Permalink
Post by evdplancke
1) there is a benefit of using 48khz instead of 44,1khz because of the
reconstruction filter
2) fanatics of NOS DAC will benefit from hires even up to 384khz
because the higher the fs, the flatter the sin x/x lowpass in audible
frequency band...
Then what about a NOS DAC without output filtering (using ear natural
lowpass filtering): this would get rid of pre-ringing filtering
artefacts. Combined with ultra hires, wouldn't it be close to the
perfect sound?
1) IME Only very few people can hear the small difference between 44.1
and 48. I used to be able to 15 years ago when DAT machines were
popular but I can't anymore. Even then it was a tiny tiny difference on
my top end Sony and I couldn't in all honesty say it was really down to
the filtering. It might just have been a better DAC.
2) All of the NOS DAC's I've heard sound uniformly dreadful in the
upper mid and top end to me
YMMVof course.
--
Phil Leigh

You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1
DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's,
ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend
Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus
Interconnect cables
Stax4070+SRM7/II phones
Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything.
------------------------------------------------------------------------
Phil Leigh's Profile: http://forums.slimdevices.com/member.php?userid=85
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
TheOctavist
2012-02-17 08:33:15 UTC
Permalink
"NOS DACS" are a ruse.

http://www.head-fi.org/t/437340/any-benefits-from-having-a-higher-sample-rate/15#post_5905139
--
TheOctavist

Vortexbox>SBT(stock)>>Forssell MDAC-2>>>Klein and Hummell 0300D

Sota Sapphire/Lyra Kleos>>Bespoke Valve Phono Stage>>Mastersound Due
Venti>>Link Audio K100
------------------------------------------------------------------------
TheOctavist's Profile: http://forums.slimdevices.com/member.php?userid=52700
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Mnyb
2012-02-17 08:58:05 UTC
Permalink
Post by TheOctavist
"NOS DACS" are a ruse.
http://www.head-fi.org/t/437340/any-benefits-from-having-a-higher-sample-rate/15#post_5905139
But they do sound "different" ;)

Amazing if the punter finds that all dac's sound's similar and then
finds the nos dac very different .

how on earth do you get to logic defying conclusion that the NOS dac is
an improvement ???
--
Mnyb

--------------------------------------------------------------------
Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x
MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3
sub.
Bedroom/Office: Boom
Kitchen: Touch + powered Fostex PM0.4
Misc use: Radio (with battery)
iPad 64gB wifi +3g with iPengHD & SqueezePad
(in storage SB3, reciever ,controller )
------------------------------------------------------------------------
Mnyb's Profile: http://forums.slimdevices.com/member.php?userid=4143
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
adamdea
2012-02-20 14:25:49 UTC
Permalink
Post by evdplancke
2) fanatics of NOS DAC will benefit from hires even up to 384khz
because the higher the fs, the flatter the sin x/x lowpass in audible
frequency band...
Then what about a NOS DAC without output filtering (using ear natural
lowpass filtering): this would get rid of pre-ringing filtering
artefacts. Combined with ultra hires, wouldn't it be close to the
perfect sound?
There is an element of truth in it if you are determined to see a point
in NOS dacs, but it is still utterly illogical.
All you are saying is that NOS dacs will work better if oversampling
(in the sense of increasing the sample rate from the native rate of the
data being decoded) is unnecessary.
Oversampling is unnecessary if the native rate of the data is already
at the rate you would have oversampled to.

A nos dac with data fs = 192k will presumably bahabve in the audible
band just like a 4x oversampling dac weith 44.1 data but no filter.

So now with no reconstrction filter and Fs = 192k or whatever we only
have to aorry about images at 96Khz plus. This does not remove the
problem that somw downstream electronics will still not like the spuria
at 96Khz. I am also not sure whether the combination of intermoudation
distortion and imaging which causes NOS dacs to have such a poor noise
floor at 44.1 will also affects them at fs =192 or whatver.

Anyway there is now no reason *not* to have a reconstruction filter.
(when i say reason I mean the spurous "reason" which justified nos dacs
in the first place). Even if there are audible pre-ringing artefacts at
fs=44.1 surely even an audiophile wouldn't think they existed at
fs=192k (which you will note is Bruno Putzeys point about apodising
filters) .

So at higher sampling rates the nos dac makes no sense whatsoever
--
adamdea
------------------------------------------------------------------------
adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
JohnSwenson
2012-02-23 08:35:21 UTC
Permalink
I have a lot to say on this subject, but its too much for one post so
I'll start with the topic of NOS DACs since its being discussed right
now.

First off a bit of background, I have been building DACs for quite a
few years, I have built at least 30 different DACs with many different
DAC chips, filters, output stages etc. I have tried hard to come up
with at least some correlations between differences in what I have been
building and what I have heard, this hasn't been easy, there are so many
variables it's hard to narrow things down.

And yes I found there ARE differences in sound, they do not all sound
the same. I have done quite a few blind tests with other people
listening, they were at least single blind, sometimes double blind. The
blind tests do prove that there are differences, but preferences as to
what people preferred went all over the place. So the following is
going to be my own personal preferences, which are certain to be
different than other people.

Before going into details of my NOS experience let me say that I am an
electronics engineer, I am well versed in sampling theory and have a
moderate acquaintance with DSP, but by no means the worlds greatest
living expert.

When I first started Building DACs I was firmly in the oversampling is
a great thing camp, it radically improves upon the old style brickwall
analog filter etc. Then I heard about NOS DACs and decided to build a
couple and see what they sound like. I was sure it was going to be
terrible. But low and behold that is not what happened. I found the
sound was significantly improved in some areas and significantly
degraded in others. This was a big surprise, I was not expecting the
improvement at all. At this point I had no clue WHY it sounded better.


I tried several NOS DAC designs of my own and several from other
people. What I found was that a lot of the NOS DAC designs "out there"
were seriously flawed in many ways, it was obvious the people designing
them really didn't know what they were doing. Frequently I could make
them sound much better by some simple changes.

Even with some of these bad design choices they all definitely have a
"NOS sound". To me the improvement part is an increase in subtle
detail, being able to hear subtle details of the acoustic environment,
being able to hear subtle nuances in performance that purvey emotional
content better. At the same time, the sound is "dirty", its rough
around the edges. When you go back to oversampling it sounds much
cleaner, but also flat and boring in comparison.

My supposition is that the people that like NOS DACs are willing to
ignore the bad parts in order to gain the good parts. It might also
have to do with age, young people with good hearing may be bothered
much more by the "dirty sound" (presumably the infamous aliasing).

I then spent a couple years trying to find out why the NOS DACs sound
better. I won't go into the full journey here, but what I eventually
concluded was that it was the digital filters themselves getting in the
way. I could build my own digital filters that kept the good qualities
of the NOS sound but without the "dirt". I wasn't doing anything
special with my filters, just a good proper implementation in an FPGA.
The only thing I can think of as to why the commercial ones do not
sound so good is that they are NOT properly implemented.

In order to properly do a brickwall filter for 44.1 takes a fair amount
of hardware resources, my guess is that the manufacturers are cutting
corners to save money. They are doing just barely enough to get decent
numbers in the data sheets. In particular I'm leaning towards the
practice of cascading several small filters rather than using one
properly implemented large filter. Looking at data sheet plots of
filter performance you can frequently see this cascading of filters.

An interesting occurrence happened early in my DAC quest (before my
first NOS DAC) I had an inexpensive DVD player and decided to see if I
could get it to sound better. I did a number of analog and PS
improvements which improved things significantly, but I also noticed
that the DAC chip had a "slow rolloff" filter setting as well as the
default brickwall filter. So I built a little board that let me
reprogram the registers in the DAC chip so I could set the filter type.
I found that I liked the slow rolloff much better. In blind tests most
people liked the slow better, but several hated it. Looking at the data
sheet plots I found that the slow filter was implemented as a single
filter but the brickwall was three cascaded filters.

I have done similar tests with many other DAC chips and with my own
digital filters and its turning out to be a fairly decent correlation
that the biggest impact is not the filter function itself but if its
implemented as cascaded filters. Filter functions DO make a difference,
but if they are all implemented as cascaded filters they all don't sound
so great. With filters implemented as single proper filters (enough
internal bits for the number of taps and enough bits for the
coefficients) differences in filter functions CAN be heard but they are
not very large. Just getting the filters implemented properly is the
biggie.

This brings up the issue of software upsampling and NOS DACs. First off
NOS does not mean no filter, just not a digital filter, you can still
put an analog filter on a NOS DAC. If the builtin digital filters are
the problem, it seems that a good NOS DAC playing upsampled files that
were generated with a properly implemented software filter should
provide good sound. And my experience is that indeed it does,
especially if you put a 2 or 3 pole analog filter after the NOS DAC to
get rid of residual high frequency noise. Note this has to be a GOOD
NOS DAC, not one of those cheap 16 bit ones from people that have no
clue what they are doing.

There are a lot of people that are doing software upsampling and
feeding the results into soundcards and external DACs that I think are
trying to do the same sort of thing, but the data is still going
through the compromised digital filter in the DAC chip. It would be
much better if they fed it through a good NOS DAC.

An interesting side bar on this is an early experiment I did. I had
been reading about people that stacked 8 1543 DAC chips, I tried this
in two different ways. One group of people spread the chips out on a
board (see the picture somewhere up in this thread), but others
actually stacked the DAC chips on top of each other. I tried both and
found the stacked on top of each other approach sounded much better.
Note this was EXACTLY the same circuit, just a different physical
layout of the chips. The difference was that the stacked chips got HOT.
Doing so cut off the airflow so they got to much higher temperatures.

I hypothesized that it was this higher temperature rather better
linearity or lower noise that made the improvement. I decided to test
this by gluing a power resistor to a single chip and pumping DC through
it to raise the temperature. I added a thermocouple so I could check the
temperature of the chip (well the temperature of the case, not the
actual chip). I then very slowly raised the temperature of the chip and
low and behold it sounded way better as the temperature went up high.
(still not all that great, the single chip by itself is a pretty bad
sounding DAC chip) So all that theory that it sounded better because of
the increased bit depth because of stacked chips was hooey, it sounded
better because it got HOT. BTW the one with the 8 chips spread out on
the board sounded worse than a single chip, that was just a bad idea.

Things HAVE been getting better. The latest crop of chips seem to not
have as bad digital implementations as previous ones (with the decrease
in cost of compute hardware, its probably cheaper to just do it right
than spend the money trying to develop creative corner cutting). That
doesn't mean they are perfect. Every one I have tried I have been able
to make sound better by disabling the internal digital filter and using
a properly implemented external filter. The only ones I have found that
seem to do a pretty good job of their internal filters are the Sabre
chips.

Well there you have it, some of my exploration of NOS DACs.

John S.
--
JohnSwenson
------------------------------------------------------------------------
JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Phil Leigh
2012-02-23 11:45:47 UTC
Permalink
So John, are you saying that on 44.1 material you prefer a NOS with a
good analogue filter set for 22Khz vs an ASRC DAC running internally at
384 or 768 with an appropriate digital filter?

Just curious.

That "dirt" you refer to is exactly what I don't like about all of the
NOS DAC's I've heard. Of course this doesn't mean ALL NOS DAC's are
empirically bad... simply that I personally haven't heard what I
consider to be a good one.

Interesting about the heat phenomena. The heat must be causing the
Johnson noise to rise inside the DAC, no:? - maybe the noise is having
a "dithering" effect?
regards
Phil
--
Phil Leigh

You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1
DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's,
ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend
Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus
Interconnect cables
Stax4070+SRM7/II phones
Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything.
------------------------------------------------------------------------
Phil Leigh's Profile: http://forums.slimdevices.com/member.php?userid=85
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Archimago
2012-02-23 16:13:36 UTC
Permalink
Thanks for sharing your experience with DAC's John!

Over the years I have played with a cheap DIY 4-chip TDA1543 design
similar to the stuff on eBay as well as a more expensive Mhdt
Constantine (TDA1545 I think). Can't say I was enamoured with the
sound... I found the highs a bit too rolled off for my taste so
happily went back to the Transporter sound.

Wondering if there was a commercially available NOS DAC you think
represents a good design. I don't have time anymore to fool around with
DIY's.

As for the slow roll-off filter, any opinions on the Transporter's
AK4396 slow-roll vs. standard filter?
--
Archimago
------------------------------------------------------------------------
Archimago's Profile: http://forums.slimdevices.com/member.php?userid=2207
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
JohnSwenson
2012-02-25 01:07:08 UTC
Permalink
Post by Archimago
Thanks for sharing your experience with DAC's John!
Over the years I have played with a cheap DIY 4-chip TDA1543 design
similar to the stuff on eBay as well as a more expensive Mhdt
Constantine (TDA1545 I think). Can't say I was enamoured with the
sound... I found the highs a bit too rolled off for my taste so
happily went back to the Transporter sound.
Wondering if there was a commercially available NOS DAC you think
represents a good design. I don't have time anymore to fool around with
DIY's.
As for the slow roll-off filter, any opinions on the Transporter's
AK4396 slow-roll vs. standard filter?
If I were going to buy a NOS DAC today I would get the Audio-gd DAC-19,
it uses a pair of 1704K chips (my personal favorites). It also does what
I do, it implements its own digital filters in an FPGA (strangely enough
exactly the same FPGA chip I use in my latest DAC). You can also set it
to NOS mode if you desire.

Its not the cheapest DAC nor the most expensive. I think it does a good
job all around (except for maybe the USB input).

I have not had a change to listen to the Transporter's slow rolloff
mode so I can't comment on that.

John S.
--
JohnSwenson
------------------------------------------------------------------------
JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
JohnSwenson
2012-02-25 01:25:14 UTC
Permalink
Post by Phil Leigh
So John, are you saying that on 44.1 material you prefer a NOS with a
good analogue filter set for 22Khz vs an ASRC DAC running internally at
384 or 768 with an appropriate digital filter?
Just curious.
That "dirt" you refer to is exactly what I don't like about all of the
NOS DAC's I've heard. Of course this doesn't mean ALL NOS DAC's are
empirically bad... simply that I personally haven't heard what I
consider to be a good one.
Interesting about the heat phenomena. The heat must be causing the
Johnson noise to rise inside the DAC, no:? - maybe the noise is having
a "dithering" effect?
regards
Phil
My preference is a DAC which uses properly implemented digital filters,
unfortunately these are rare. The NOS DAC and software approach is a way
to get around trying to find hardware with good digital filters. By
doing the oversampling in software you have FAR more flexibility in
playing with different filter parameters.

For my own DACs I've spent years going back and forth between NOS and
digital filters. A NOS DAC with several poles of analog filtering can
sound quite good. But the best is still digital filtering done right.
Of course that takes significant digital horse power, which means you
have to be very carfeful that the noise from the filter doesn't make it
into the clock circuits, DAC chips and analog circuits. That takes very
careful grounding and PS design, none of which comes cheap.

I personally do not like ASRCs. I have not heard a single one that I
really like. Of course it may not be the ASRC per se. I expect part of
it might be the interaction of the imperfect digital filters in both
the ASRC and the DAC chip interacting with each other.

On the "HOT DAC" I suspect its due to internal PS noise. As the chips
get hotter the FETs in the circuits get higher resistance, which slows
them down (RC goes up) so the peak current from each switching goes
down, thus less noise on power and ground traces in the chip and
package. As the temperature goes up at some point things get so slow it
stops working, you don't want to go that far!

John S.
--
JohnSwenson
------------------------------------------------------------------------
JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Mnyb
2012-02-25 04:15:37 UTC
Permalink
The trend I was referring to with my blanket dismissal of NOS DAC was
the ones completely without any filter, this was very popular as DIY
for while so popular that it seems to be the norm :-/
--
Mnyb

--------------------------------------------------------------------
Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x
MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3
sub.
Bedroom/Office: Boom
Kitchen: Touch + powered Fostex PM0.4
Misc use: Radio (with battery)
iPad 64gB wifi +3g with iPengHD & SqueezePad
(in storage SB3, reciever ,controller )
------------------------------------------------------------------------
Mnyb's Profile: http://forums.slimdevices.com/member.php?userid=4143
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Wombat
2012-02-26 02:13:31 UTC
Permalink
Hi Mnyb i just realized today that your signature must have changed
lately. IsnŽt it?
Using new Meridian DSP is like cheating! No wonder you donŽt hear a
difference with transports. These babies simply seem to work :)
Congrats for the purchase!
--
Wombat

Transporter (modded) -> RG142 -> Avantgarde Acoustic based 500VA
monoblocks -> Sommer SPK240 -> self-made speakers
------------------------------------------------------------------------
Wombat's Profile: http://forums.slimdevices.com/member.php?userid=4113
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Mnyb
2012-02-26 10:37:18 UTC
Permalink
Post by Wombat
Hi Mnyb i just realized today that your signature must have changed
lately. IsnŽt it?
Using new Meridian DSP is like cheating! No wonder you donŽt hear a
difference with transports. These babies simply seem to work :)
Congrats for the purchase!
Yea I hesitated for years before I did it they are are not perfect but
they are done right, makes other hifi looks like steam engines (nice
and complicated and well put together but why ? )

They are a good proof of concept how hifi and home theater should be
done .
A pity they are the only game in town :-/ if others "get it" (including
other brands and audiophiles ) this kind of solution would go down in
price and improve at the same time .

I'm a tech nerd :) I do like the underlying architecture of this
system, I wish meridian could license their speaker-link MHR connection
. Or if someone could make a conection standard for digital active
speakers.

I have used the processor much longer than the speakers.

It all started when DVDA was hot , there where not many that offered
6ch 24/96k digital link from the player to the processor (now with hdmi
everyone has it ) .
The benefit of 24/96 may be questionable , but using an analog DA-AD-DA
route via a traditional HT receiver will blow even that small chance to
improvement away .

This system is source transparent even trough the xover and sound
degradation does not start until the the signal Reach the DAC and amp
for that speaker driver .

Strictly I could have bought some nice analog active speakers and maybe
get better sound. But the Meridian solution is so neat just cat5 cables
to the speakers
--
Mnyb

--------------------------------------------------------------------
Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x
MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3
sub.
Bedroom/Office: Boom
Kitchen: Touch + powered Fostex PM0.4
Misc use: Radio (with battery)
iPad 64gB wifi +3g with iPengHD & SqueezePad
(in storage SB3, reciever ,controller )
------------------------------------------------------------------------
Mnyb's Profile: http://forums.slimdevices.com/member.php?userid=4143
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
darrenyeats
2012-02-26 12:17:51 UTC
Permalink
Post by Mnyb
Strictly I could have bought some nice analog active speakers and maybe
get better sound. But the Meridian solution is so neat just cat5 cables
to the speakers
Although personally I like ATC (not just the sound but the ethos) I
heard Meridian DSP5200 a few years ago at a show and I thought they
sounded great and are, as you say, an extremely neat and modern
solution. Good choice IMO.

I heard the M80 at the same show and I didn't like it, TBH. But I
wouldn't mind a listen of the DSP3200 plus DSW sub...!
Darren
--
darrenyeats

http://www.amazon.co.uk/gp/richpub/listmania/byauthor/A3H57URKQB8AQO/ref=cm_pdp_content_listmania/203-7606506-5721503.

SB Touch
------------------------------------------------------------------------
darrenyeats's Profile: http://forums.slimdevices.com/member.php?userid=10799
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Mnyb
2012-02-26 14:35:22 UTC
Permalink
Post by darrenyeats
Although personally I like ATC (not just the sound but the ethos) I
heard Meridian DSP5200 a few years ago at a show and I thought they
sounded great and are, as you say, an extremely neat and modern
solution. Good choice IMO. Active speakers have a lot of advantages and
of course Meridian have their own unique take even on the active
architecture.
I heard the M80 at the same show and I didn't like it, TBH. But I
wouldn't mind a listen of the DSP3200 plus DSW sub...!
Darren
Yea ATC been around since the 70's and Meridians first DSP speaker
arrived 1990 22 years ago ? And no one else picked it up ? :-/ I may
not repeat what I think of the so called "design" of modern high end.
--
Mnyb

--------------------------------------------------------------------
Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x
MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3
sub.
Bedroom/Office: Boom
Kitchen: Touch + powered Fostex PM0.4
Misc use: Radio (with battery)
iPad 64gB wifi +3g with iPengHD & SqueezePad
(in storage SB3, reciever ,controller )
------------------------------------------------------------------------
Mnyb's Profile: http://forums.slimdevices.com/member.php?userid=4143
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
TheOctavist
2012-02-27 04:13:17 UTC
Permalink
my klein and hummel 0300d with the c28 has loads of room correction
options..of course they are also digital
--
TheOctavist

Vortexbox>SBT(stock)>>Forssell MDAC-2>>>Klein and Hummell 0300D

Sota Sapphire/Lyra Kleos>>Bespoke Valve Phono Stage>>Mastersound Due
Venti>>Link Audio K100
------------------------------------------------------------------------
TheOctavist's Profile: http://forums.slimdevices.com/member.php?userid=52700
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Mnyb
2012-02-27 05:26:25 UTC
Permalink
K&H is a pro monitor speaker very neat design, the xover is analog thou,
transformer coupled analog input ?

But you are using an old forsell DAC not the K&H built in DAC ?

If you want to use the digital input how do you practically do things
like adding a subwoofer and how do you do hometheater , using the
analog inputs ? Or a pro mixing console or software and multi channel
sound card ?

The 300 seems to be a control desk monitor do they work fine on stands
used as normal speakers, they got one very good feat, centre channel
would be identical .

I reckon K&H deemed the home use Market hopeless :-/ not any models
adapted to that.
--
Mnyb

--------------------------------------------------------------------
Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x
MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3
sub.
Bedroom/Office: Boom
Kitchen: Touch + powered Fostex PM0.4
Misc use: Radio (with battery)
iPad 64gB wifi +3g with iPengHD & SqueezePad
(in storage SB3, reciever ,controller )
------------------------------------------------------------------------
Mnyb's Profile: http://forums.slimdevices.com/member.php?userid=4143
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Mnyb
2012-02-27 05:56:55 UTC
Permalink
how do one controll the volume of a pair of digitally connected K&H ?
Example in my meridian system volume controll is in the speakers and
the "volume information" is passed on from the processor in the speaker
link .
--
Mnyb

--------------------------------------------------------------------
Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x
MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3
sub.
Bedroom/Office: Boom
Kitchen: Touch + powered Fostex PM0.4
Misc use: Radio (with battery)
iPad 64gB wifi +3g with iPengHD & SqueezePad
(in storage SB3, reciever ,controller )
------------------------------------------------------------------------
Mnyb's Profile: http://forums.slimdevices.com/member.php?userid=4143
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
arnyk
2015-05-28 06:59:33 UTC
Permalink
Post by Mnyb
The trend I was referring to with my blanket dismissal of NOS DAC was
the ones completely without any filter, this was very popular as DIY for
while so popular that it seems to be the norm :-/
That's it in a nutshell.

Most recent work with NOS DACs has centered around the Philips TDA 1541
as a more-or-less stand alone DAC with no or minimal to vestigial low
pass filtering. If you study its history it was never really intended by
its developers to be used that way. Information about how the TDA 1541
was supposed to used was discussed in the early days of digital audio
(1981-1983) by professional magazines like as EDN and EE Times (if they
existed them and if they didn't their forebearers), but issues of those
periodicals that old seem to be lost to posterity. There may be relevant
Philips Application notes, but I can't find them either.

The TDA 1541 chip was originally designed to be a component of an
oversampled DAC, most significantly to be followed by a digital filter
chip such as the SAA7030 and its sequels. One fact that I haven't seen
is references to what used to be called Aperture Effect which is the
reason why the output of many DAC chips have an unexpected (to many)
droop at high frequencies. This issue is discussed in the context of
data acquisition here: http://www.cypress.com/?docID=45630.

When used all by itself the output of a TDA 1541 chip is going to have
that HF-drooping frequency response that is shown in the Cyprus
Semiconductor reference in figure 1. The digital filter chips such as
the SAA7030 that it was supposed to be used with such as the contained
corrections for this that were in the day called . It is as simple as
that - the TDA 1541 was never intended to be used without some kind of
Aperture Correction, and if you don't provide it, its going to sound a
little soft compared to an accurate DAC.

There are other potential bad consequences to not having an appropriate
reconstruction filter following the DAC such as IM in following stages
of amplification due to the fairly large amounts of HF noise that is not
filtered out like it should be. The noise itself is > 22 KHz and likely
to not be heard by many if anybody. The IM doesn't always happen to an
audible degree, but I'm sure given the relatively high amplitude of this
noise it does happen some times. I suspect that there are even people
who like their music with a little aharmonic nonlinear distortion
spurious responses tossed in.


------------------------------------------------------------------------
arnyk's Profile: http://forums.slimdevices.com/member.php?userid=64365
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
adamdea
2012-02-23 17:29:37 UTC
Permalink
Post by JohnSwenson
This brings up the issue of software upsampling and NOS DACs. First off
NOS does not mean no filter, just not a digital filter, you can still
put an analog filter on a NOS DAC. If the builtin digital filters are
the problem, it seems that a good NOS DAC playing upsampled files that
were generated with a properly implemented software filter should
provide good sound. And my experience is that indeed it does,
especially if you put a 2 or 3 pole analog filter after the NOS DAC to
get rid of residual high frequency noise. Note this has to be a GOOD
NOS DAC, not one of those cheap 16 bit ones from people that have no
clue what they are doing.
There are a lot of people that are doing software upsampling and
feeding the results into soundcards and external DACs that I think are
trying to do the same sort of thing, but the data is still going
through the compromised digital filter in the DAC chip. It would be
much better if they fed it through a good NOS DAC.
John this is fascinating stuff, but the terminology is getting
confusing to me. NOS literally stands for non oversampling (?) - but
tends to be associated with either no filter or analogue output filter.


Forgive my ignorance but how do "upsampled files that were generated
with a properly implemented software filter" differ from the
up/oversampling which would go on in an ordinary OS dac? It seems to me
that you clearly have both oversampling and a digital filter here- your
just doing it in software.


For the sake of simplifying Inguz all my files are (courtesy of Phil's
tweak) upsampled by sox to 96 kHz before streaming to my dac. Is this
what you have in mind? I can't see how you can go beyond 96kHz via a
squeezebox

If the processing power in the dac is the limiting factor I see why it
might be better to finesse this in software. I hope I am not making a
fool of myself by I can't see how this differs in principle from the
classic OS dac with a final analog filter to remove images above half
the oversampling frequency. Would this be NOS in the sense that
Kusunoki meant it http://www.sakurasystems.com/articles/Kusunoki.html

Presumably with will only work with a multi bit dac this natively works
at the upsampled frequency not a delta sigma one as those will
presumably inevitably have their own filters.

[I fear I am betraying my ignorance but I have never been able to grasp
the difference between upsampling and oversampling]
--
adamdea
------------------------------------------------------------------------
adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603
View this thread: http://forums.slimdevices.com/showthread.php?t=93483
Loading...