MEASUREMENTS: Some Squeezebox numbers to consider...

[Archimago]

Hey guys, as I'm waiting for the final tally on the MP3 test to come in,
I thought I'd try my hand over the last week at running some RightMark
Audio Analyzer 6.2.5 (most recent version from July 2012) tests. Around
Christmas, I got my new ASUS Xonar Essence 1 as my main DAC so as a
result had the old E-MU 0404USB (drivers suck and headphone amp too weak
for AKG Q701) sitting around. I figured I'd try to put it to good use
with some measurements and see what comes up.

What I can say so far is that RMAA is remarkably consistent with good
inter-test reliability so long as one has the technique figured out with
one's equipment. It's very easy to get things wrong during calibration
for example. As usual, there are the caveats to keep in mind and the
results I get here may not be comparable with others (good writeup here:
http://nwavguy.blogspot.ca/2011/02/rightmark-audio-analyzer-rmaa.html).

Here's the general setup:
- Computer: Intel i7-3770K @ 4GHz, 24GB DDR3
- E-MU 0404USB --> USB2 of the computer motherboard using generic USB
cables. E-MU located about 3 feet from the main computer (too close
results in obvious RF interference raising noise floor). For 16/44
testing I record at 24/44, and 24/96 for 24/96.
- For unbalanced testing: RCA (3ft, Radio Shack shielded cables ~$10)
--> XLR adaptors (maybe $25 for the set)
- For balanced testing: generic XLR cables (bought at pro music shop
years back, maybe $20)


First up: ASUS Essence 1
A few observations. Well made DAC. Likely much more expensive if came
from some boutique audio brand. Heavier and larger thank I expected!
Should be a good test of the approx limits of the E-MU as testing gear.

24/96 signal:
14302

The first 2 columns are the result between using the RCA vs. XLR cables.
Clearly XLR is the way to go with 5dB improvement in noise floor,
dynamic range, and stereo crosstalk. Bottom line, XLR buys you about
1-bit in digital resolution according to my setup.

The last 2 columns are with the computer running full load (hence the
note "NOISY i7). I was running the machine full tilt with Prime95 and
the graphics card inside (nVidia GTX 570) doing the "Furry Cube"
demo/bechmark from GPU Caps Viewer. Notice how the RCA unbalanced setup
resulted in significantly increased noise floor by 14dB (about loss of 2
bits resolution)! XLR setup however ran stably with no evidence of the
noise pollution.

In theory I would have expected that a busy machine worsens "jitter"
through the USB interface with all the audiophile talk about minimizing
# threads and CPU load... At least no evidence from my tests to suggest
this makes any difference SO LONG AS YOU MINIMIZE RF NOISE from getting
into the system.

Frequency response (no real issue):
14303

Noise floor:
14304

Clearly the RCA NOISY spectrum is much inferior to the others (cyan).
However, the RCA quiet computer (white) has a number of spikes as well
evident.

CONCLUSION (assuming you believe RightMark's methodology and my
competence :-):
1. My XONAR - E-Mu system is capable of measuring down to ~113dB
(>18-bits) noise floor.
2. If you have a choice, XLR (balanced) beats RCA (unbalanced) hands
down!
3. If you're listening with the computer, computer LOAD makes a
difference but if your equipment has good noise rejection (eg. use of
balanced cables), this does not seem to be an issue... If computer load
correlates to jitter, I'm not seeing it showing up in these
measurements.


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[Archimago]

Next up - my classic "Slim Devices" Squeezebox 3 (I believe this is one
of the "first run" units; I was on a waitlist at introduction):

RESULTS:
14305
First 2 columns are the stock SB3+stock wallwart connected to my
basement music server by WiFi. Notice that 24-bit data does result in
dropping of the noise floor by ~6dB. It looks like the good ol' SB3
internal DAC is capable of about 17-bit resolution when fed with
24-bits. Note that the old Stereophile review from 2006 did not measure
24-bit performance.

Second 2 columns are the same setup but with the ethernet (hooked up to
my DLink gigabit switch 6 feet away). Essentially no difference compared
to the WiFi.

Final column is with the SB3 over WiFi but the *SB Touch wallwart*. I
see folks here talking about the crappy wallwart (true, the UNIFIVE
wallwart looks and feels nasty compared to the one that came with the
Touch!). I fed the 24-bit data and the result is essentially the same as
the UNIFIVE. Based on these measurements, I don't see any evidence that
the Touch wallwart would improve the stock SB3 performace.

Frequency Response:
14306

Decent - obviously not as flat as Essence 1 above from 20Hz-20kHz...

Noise Floor:
14307

16-bit data obviously not as good as 24-bits. No difference between the
WiFi vs. Ethernet groups.

CONCLUSION:
1. The SB3 can benefit from 24-bit "hi-res" audio. Whether you can hear
the extra 5-6dB is your problem :-)
2. I see no evidence that a Touch wallwart would improve the performance
over the cheap stock power supply.
3. No evidence that running in WiFi mode will add any noise to the SB3
output. (Also, no evidence of higher WiFi data latency or possible
jitter issues showing up in these measurement.)

Next up... SB Touch this afternoon...


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[ralphpnj]

Interesting, very interesting!

Thanks for this very useful and insightful information. Please keep up
the good work.


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[Archimago]

ralphpnj wrote:
> Interesting, very interesting!
>
> Thanks for this very useful and insightful information. Please keep up
> the good work.

Oh, the things we do for a hobby :-)
It's fun to try this stuff to hopefully understand it myself instead of
getting spoon-fed by the standard audiophile press.

I added the THD graphs as well to the above.

Along with the RightMark measurements, I recorded the Dunn J-Test signal
through the devices also but am having a heck of a time figuring out
what FFT program & parameters to use (eg. what FFT size,
Hanning/Hamming/Blackmann..., etc.) for jitter analyze. So far, the
signals look very clean; it's certainly possible the E-Mu's ADC jitter
is high and overwhelming the picosecond effects. Anyone in the know
about the J-Test please point me to the right direction!

Some might wonder if a better USB cable makes a difference... Short
answer: NO.

Long answer:
I torture-tested the USB2.0 connection by:
1. Connecting the E-Mu in loopback (both DAC & ADC functions) mode to
the front USB connector which is attached to the motherboard USB header
with thin unshielded wires.
2. I connected the E-Mu in loopback to an external USB 7-port hub.
3. While in condition 2, I did a file copy with an external USB
harddrive transferring ~24MB/sec through that hub while running the
RightMark test!

Even under condition 3 which is about as bad as you can get in terms of
saturating the USB port, there was no difference in the results! Usual
charts and graphs below...
14313
14314
14315
14316

And a picture of the SB3 on my test bench this AM getting checked out
:-)
14317


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[Archimago]

As promised, here's the SB Touch measurements... I got this guy late
last year lightly used when Logitech announced the Touch's
discontinuation:

RESULTS:
14319

Clearly better results all around compared to the SB3. Frequency
response within tighter range over the 20-20kHz spectrum, noise levels 1
dB lower in the 16-bit domain (as if it could go any lower!?), and about
-4dB lower with 24-bit data giving the Touch DAC about 17.5 bits of
dynamic range. Interestingly, the stereo crosstalk is a bit higher in
the Touch vs. SB3 by about 6-8 dB... Interesting.

Notice no difference between WiFi and wired through the ethernet.

Frequency Response:
1432014323

16-bit data left, 24-bit right. Looks good. Less bass drop-off than the
SB3 with my equipment.

(to be continued - could not attach more pix)


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[Archimago]

Noise Floor:
1432614327

THD Graph:
1432814329

Summary:
I guess the only surprise is that the stereo crosstalk is higher in the
Touch than the SB3. Otherwise, audio quality seems superior - it can
obviously handle 96kHz natively (up to 192kHz with the EDO plug-in), and
flatter frequency response especially in the low bass could be
noticeable.

Again, assuming the WiFi strength is reliable and you're not constantly
rebuffering, I see not indication that sound quality is negatively
impacted by going wireless.

Later this coming week... The Transporter. BTW, I also have a Boom and
Radio. Let me know if anyone wants to have a peek at the analogue
outputs.


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[Mnyb]

Extremely cool :) seems like good software ,but you must have a good
soundcard for yourpc before it gets usefull ?

Did you try any of the software tweaks on your Touch it would be
interesting to know if the output is changing at all .


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[Archimago]

Mnyb wrote:
> Extremely cool :) seems like good software ,but you must have a good
> soundcard for yourpc before it gets usefull ?
>
> Did you try any of the software tweaks on your Touch it would be
> interesting to know if the output is changing at all .

Haven't tried any of the software tweaks yet on the Touch... Any
suggestions of maybe the most extreme tweak(s) I should try to see if I
can measure a difference?


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[Mnyb]

Archimago wrote:
> Haven't tried any of the software tweaks yet on the Touch... Any
> suggestions of maybe the most extreme tweak(s) I should try to see if I
> can measure a difference?

I have not realy used any of them seriously I only use the digital out
in my serious system and on the kitchen system I don't care :)

just installing TT3 with it's defaults , implement all iterations of
weird settings used by it followers would be an endless task.

But the most weird thing is they suggest that blocking the volume
setting fro the toolbox sounds different than doing it from LMS or just
keep the volume at 100% :)


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[Archimago]

I got the AUNE X1 last year as a "portable" headphone amp / DAC. I did
an AB test using this with a friend's Weiss DAC202 (massive price
differential) and it sounded pretty good in comparison :-).

14334

The USB implementation is the bog standard BB PCM2707 (adaptive
isochronous, only up to 16/48 sampling rate USB 1). The internal DAC is
the BB PCM1793.

Supposedly, these non-asynchronous USB DAC's are prone to jitter so I
wanted to see what happens with the USB hub + external HD file copy
condition:
14339

So using the 7-port USB hub, I plugged in the E-MU 0404, the AUNE X1,
and the USB hard drive (LaCie, about 3 years old).
On the left is the quiet USB hub - no activity other than the test being
run - audio played with the AUNE X1 --> 0404 for measuring. Not bad;
pretty ideal 16-bit numbers.
In the middle column - I started a massive file copy from the USB HD -->
internal HD while recording the test (avg transfer rate ~25MB/sec). No
difference.
On the right, I tortured the hard drive with a file copy to itself - ie.
both read and write to the same USB HD (avg transfer rate ~12MB/sec).
Notice there IS a rise in the noise floor by 3 dB.

Frequency response:
14340

Flat. No difference.

Noise level:
14341

As you can see, in the USB HD "self copy" condition, there is an
increase in the noise floor. Notice that the noise hits the lower
frequencies primarily.

THD:
14342

Finally for the THD measures, again, the deviation is primarily in the
lower frequencies although there are some distortions around the 1kHz
signal as well.

Conclusion:
Not exactly sure what to make of this really, but I'll give a
suggestion. I guess the bottom line is not to have excessive data
transfer through the USB bus when listening with a USB DAC (no
kidding...). Having said this, I think what I'm seeing here is noise
from the external USB HD polluting the DAC rather than any kind of
jitter explanation. I suspect that when the HD is used for writing, the
electrical noise level goes up and that's all I'm seeing. This would
explain the noise floor increasing in the lower frequencies. So, perhaps
the better conclusion is to make sure noisy peripherals are not plugged
into the USB system when you're listening to the DAC. In case you're
wondering, I did this test 3 times in each condition and the rise in
noise with the HD 'self copy' case was consistent.

Hmmm, if I can find some time tonight, will give the Touch TT3 software
mod a try :-). I just don't want to brick my precious Touch!


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[Mnyb]

Is the headphone amp USB bus powered ?


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[Archimago]

Mnyb wrote:
> Is the headphone amp USB bus powered ?

No, the AUNE has its own wallwart. I used the RCA out on the back for
the measurements.

Any Touch TT3 modders reading this? Is it easy to go back to stock
settings afterwards?


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[Mnyb]

Archimago wrote:
> No, the AUNE has its own wallwart. I used the RCA out on the back for
> the measurements. I bet you something like the Audioquest Dragonfly
> would measure even worse with the USB noise, asynchronous or not I
> suspect is irrelevant. Dont know if reviewers have commented.
>
> BTW any Touch TT3 modders reading this? Is it easy to go back to stock
> settings afterwards?

Just factory reset the whole thing hold the little button above the
power conector for a while .


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[Mnyb]

There are two use cases .

As digital transport or listening via analogue out .

There is some merit to that something could happen on the analog outs
not so likely that a connected dac exhibits differences when using it as
transport .


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[Archimago]

Mnyb wrote:
> There are two use cases .
>
> As digital transport or listening via analogue out .
>
> There is some merit to that something could happen on the analog outs
> not so likely that a connected dac exhibits differences when using it as
> transport .

Good point Mnyb:
I'll look into doing some Touch-as-transport comparisons with coaxial
vs. Toslink, EDO with analogue outputs on/off.


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[Archimago]

As mentioned in the 1st post, I'm waiting for the results of my blind
MP3 test...

In the interim, I wondered if my measurement setup can be fooled by MP3.
So, with the AUNE X1 still connected, lets run the test signal lossless
vs. MP3 through and see!

Summary results:
14345

Not bad! When condensed into a single number, the different MP3
bitrates look fine! However, here's an example where single digit
measurements can be very deceiving.

Frequency response:
14346

Again, MP3 looks pretty decent. However, you can start seeing the lower
bitrates drop off in the high frequency prematurely beginning with 128
kbps (green) in sequence. Noticeable effect with lossless compression
but not bad...

Noise:
14347

Once again, not bad... What can I say, silence doesn't need many bits
to encode :-)

THD:
14348

The "money shot" right there! It's obvious we can now differentiate the
lossless (white) from the lossy compressed. There appears to be a large
gap between the distortion from 128 to the rest. 192, 256, and 320 kbps
all congregate together pretty close and I suspect for many of us, this
correlates to 192 kbps+ being quite transparent.

This is why MP3 compression effect is probably quite close to VERY
SEVERE jitter distortion (? at least tens of nanoseconds) in that it's
correlated to the signal itself beyond simply single sidebands.

Conclusion:
You cannot fool machines with psychoacoustics!!!


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[darrenyeats]

Interesting for me, in that I decided a while ago that 192 VBR was my
line in the sand for mobile use.
Darren


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[Archimago]

darrenyeats wrote:
> Interesting for me, in that I decided a while ago that 192 (VBR) MP3 was
> my line in the sand for mobile use.
> Darren

Yup. 192 VBR should be fantastic!

Year ago, before the true VBR algorithm in LAME, I decided on 210 kbps
with the old ABR algorithm when I first ripped my CD's. Of course, over
the years, they've almost all been re-ripped to FLAC.


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[SoftwireEngineer]

Archimago.. this is lot of work !!! Great contribution to the board.
Thx. Looking forward to see any difference in DAC output when the Touch
is used as a transport with some mods applied (like TT3.0, Linear Power
supply etc).


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[Archimago]

SoftwireEngineer wrote:
> Archimago.. this is lot of work !!! Great contribution to the board.
> Thx. Looking forward to see any difference in DAC output when the Touch
> is used as a transport with some mods applied (like TT3.0, Linear Power
> supply etc).

Thanks SoftwireEng! Even though I've spent quite a number of hours on
this, it's been very enjoyable and enlightening as well! It feels like
I'm finally able to answer questions for myself which I've always wanted
to learn / know / test.

There's also a sense of "learning to fish" rather than being fed by what
many here feels amount to little more than propaganda in much of the
mainstream press...

I'll see about giving the TT mod a look later this week but am concerned
that already I'm seeing the Touch functioning almost ideally with 16/44
& 24/96 signals! I'm wondering if SoundCheck did any objective testing
when he developed the mods as I'm unclear what I should be looking
for...


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[Archimago]

Okay guys, time to talk 24/192 with the Touch with EDO plugin. Thank you
Triode - amazing plugin/kernel!

Results:
14365

What I find impressive here is the fact that TosLink to the Essence 1
worked at 24/192!!! In fact, that picture of the Touch in Part I playing
the 24/192 John Coltrane's "Blue Train" was through the TosLink (you can
see the 192kHz LED lit up on the Essence 1). This is why I'm very
impressed by the components used in the Touch; kudos to Logitech and
ASUS! Over the years, this is the first time I've been able to play
24/192 for hours without obvious clicks/pops/disruptions even with an
inexpensive plastic cable. To show the TosLink result above wasn't a
"one-off", here's a series of 4 runs with the TosLink done over about 2
hours - notice the inter-test reliability.

14364

Like with the 24/96 tests, the RCA cable is showing its limits even more
with measurably increased noise floor (10 dB worse!) from the lack of
shielding and possibly data errors as the speed of data transfer
doubles. Another interesting phenomenon is that even the shielded
coaxial cable has a measurably higher noise floor compared to the
TosLink by about 2dB! Would an expensive coaxial cable improve this?
Possibly if the shielding is good.

Another observation with this test is that the noise level, dynamic
range, and stereo crosstalk are all WORSE than 24/96. Every piece of
equipment here from the E-Mu, to Essence 1, to Touch are running at max.
specifications so the system is running at it's limit. Furthermore, I
suspect there may be issues with the RightMark benchmark itself. For
example, it doesn't seem capable of measuring the frequency response
beyond 50 kHz even though I know from other tests that I can play an
80kHz signal from the Touch -> Essence 1 -> E-Mu and successfully record
the signal (I don't think this limitation would affect the other
measurements though).

Here's the THD graph showing the increase in noise with the coaxial
interface - el cheapo RCA is looking bad here:
14366

Conclusion:
1. Thanks again to Triode for the EDO plugin/kernel. It works
beautifully and in my system with TosLink working, the Touch measures
within 1-2 dB in terms of noise level, dynamic range, and stereo
crosstalk compared to a direct USB connection to the DAC for 24/192
playback!

2. If you can get the TosLink to work, you've set yourself free from
electrical noise with "galvanic isolation" of the Touch and DAC. Again,
the fact that I could get TosLink 24/192 to work reliably between the
Touch and Essence One really is impressive and speaks well of the
equipment.

3. A recurring theme in these tests is that of ELECTRICAL NOISE. Coaxial
cables need good shielding! Like I said before, I'm not sure what to
look for to know if jitter is a problem, but it's *very* obvious in the
measurements when noise creeps in and deteriorates the noise floor.

4. 24/192 does not measure as well in my system as 24/96. Writers like
Lavry and xiph.org have already eloquently documented their opinions
against 24/192 and I guess I can echo their concerns with the gear I'm
using for these tests... Firstly, between 24/96 and 24/192, the
difference is ultrasonic; do we demand high-end SLR digital cameras to
also capture ultraviolet light? (Sure, you might want to do this for
specific scientific reasons.) Secondly, other than a handful of albums
usually from smaller labels like 2L, Reference and Linn, I have rarely
come across truly native 24/192 (or 24/176) recordings. IMO, it also
makes no sense to buy stuff like DSD64 converted to 24/176 such as many
of the HDTracks offers.

That's all for now... Probably take a break from all this testing for
the next few days :-).


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[ralphpnj]

Thank you Archimago. Truly amazing work. Very entertaining and highly
enlightening. What I take away from all these test results is that
digital audio is really quite different from analog audio and to keep
insisting on applying the same belief system, e.g. heavy duty and
expensive cables instead of well made less expensive cables, to digital
audio is never going to improve the sound. Improving digital audio
requires a different understanding and different ideas so the sooner the
audiophile understands this the sooner improvements in digital audio
will happen.

Thanks again for helping to enlighten all of us.


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[Mnyb]

Truly excellent work !

The USB E1-16/44 shows im distortion at 0.121% you got a similar result
earlier In the analog measurements , is this an anomaly ? Im distortion
is not nice .

I think it it is natural if the bandwith gets wider the noise levels
goes up try 24/48 for kicks .
The limit could also be the EMU board rather than the DAC or both .

Did I miss something is any of these measurements without EDO ? Wonder
how an of the shelf Touch performs .

While having such excellent equipment aviable , you could for kicks try
another type of listening test , I've read it in a pro audio forum I guy
was out to cure his " gear anxiety " if his ADC and DAC's of his
soundcard was reasonably transparent and if he should push anymore $$
that way.

Loop input and output and record it then you can listen to the DA - AD
chain ( Touch ,Asus , EMU ) you can do it 1 and say 10 generations ,
note very important to keep the levels reasonable and get unity gain ,so
it does not clipp or you get different levels .
( it would be to easy to spot if they got problems like that ). ( use 24
bit )

You could do fun things like listening to a file that have experienced
one ( or more ) generations of DA trough the Touch and then back trough
the EMU on your Asus trough your best hifi .

I wonder how many golden ears out there there is that actually could
tell them apart :) on thier 30k$ DAC !


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[Archimago]

Mnyb wrote:
> Truly excellent work !
>
> The USB E1-16/44 shows im distortion at 0.121% you got a similar result
> earlier In the analog measurements , is this an anomaly ? Im distortion
> is not nice .
>
> I think it it is natural if the bandwith gets wider the noise levels
> goes up try 24/48 for kicks .
> The limit could also be the EMU board rather than the DAC or both .
>
> Did I miss something is any of these measurements without EDO ? Wonder
> how an of the shelf Touch performs .
>
> While having such excellent equipment aviable , you could for kicks try
> another type of listening test , I've read it in a pro audio forum I guy
> was out to cure his " gear anxiety " if his ADC and DAC's of his
> soundcard was reasonably transparent and if he should push anymore $$
> that way.
>
> Loop input and output and record it then you can listen to the DA - AD
> chain ( Touch ,Asus , EMU ) you can do it 1 and say 10 generations ,
> note very important to keep the levels reasonable and get unity gain ,so
> it does not clipp or you get different levels .
> ( it would be to easy to spot if they got problems like that ). ( use 24
> bit )
>
> You could do fun things like listening to a file that have experienced
> one ( or more ) generations of DA trough the Touch and then back trough
> the EMU on your Asus trough your best hifi .
>
> I wonder how many golden ears out there there is that actually could
> tell them apart :) on thier 30k$ DAC !

Thanks for the comments and suggestions Mnyb.

I agree, there's something not quite right about that 16/44 measurement
of IM distortion with the E1 via USB. I tried it a few times and
although the result varies, it remains high (ranging 0.06-0.12). Since
it measures OK with the Touch as transport, I assume there must be
something strange in the USB interface/driver causing the result
especially since it seems fine at 24/96! Hmmm, will have to look around
and see if there may be known driver issues.

The Touch measurements earlier are all with the EDO firmware installed
(a 'true' audiophile needs it ;-) just that one set is at default
settings where both digital & analog outputs are active, and the other
with the special EDO "Digital Only" settings with analogue off but
capable of 192 (setting controlled in the 'Advanced' settings menu).

Since I still have the TosLink connected, I uninstalled EDO - here are
the stock TosLink numbers 16/44 & 24/96:
14367

Essentially the same as with EDO installed. I'll clarify the earlier
post.


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[Archimago]

As promised, here's some data on my Touch with Soundcheck's Touch
Toolbox 3.0
(http://soundcheck-audio.blogspot.ca/2011/11/touch-toolbox-30.html)
software mod applied.

Before I start, I just want to say that over the years I have not been
involved in any of the discussions around this mod or the merits of it.
Even though I do not necessarily agree with many of Soundcheck's
comments on his site, I do appreciate his work in putting it together
and creating the install script which was easy to run; always good to
have hobbyists experimenting with stuff and it's interesting to see the
feedback from users.

I followed his instructions to turn off the plug-ins, change to
server-side decoding of FLAC in the "File Types" tab, "No Volume
Adjustment", etc... As per instructions, WinSCP used to transfer the
script and PuTTY for logging into the Touch.

I downloaded the script with 'wireless LAN deactivated' but noticed in
the status screen that WiFi was labeled as "enabled" still, so I used
the 'tt -w' to turn off the WiFi status for good measure during a few of
the measurements. Here's the screenshot of the "status" output for the
"NoWiFi" condition in my charts:
14387

I also wanted to test the analogue output with "tt -o 1" but I could not
get analogue output to work even though the status screen said it was
enable. If anything, one would expect that turning off the screen,
wifi, server side decoding may drop the analogue noise floor maybe a few
dB's... Oh well, I guess anyone who would go through this amount of
modding would not be using the internal DAC anyways.

Procedure same as tests in previous posts, here's the result from the
Touch+TT3 --> ASUS Essence 1 --> XLR cable --> E-Mu 0404USB:

16/44:
14383

16/44 THD graph:
14384

24/96:
14385

24/96 THD graph:
14386

The first column of each table is the stock Touch using the coaxial
output (proper shielded coaxial cable of course).

Conclusion:
I see no difference with the Touch as digital transport despite screen
off, no WiFi, server side decoding on these objective measurements.
Subjectively, likewise, the Touch sounds great played through the
Essence 1... Loving the sound of some Depeche Mode as I'm typing here
with the mod still installed. I can see how not having the screen on and
the ability to "touch the Touch" to make track selections would annoy me
after awhile :-).

At least in my case with this specific DAC, the Touch Toolbox 3.0 mod
made no difference to measurements even down to below -110dB noise
floor.


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[ralphpnj]

Thanks again. Please understand that you are now damned for all eternity
to the fires of hell, as are all audiophile heretics. Oh well, them's
the breaks.


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[SoftwireEngineer]

oh my goodness, I must be deluded :-) .. You guys need to come over to
my place and blind test me :-)
Anyways, this is what Stereophile has to say -
"There is no consensus about what levels of jitter in a digital
product's output are acceptable—the audibility will depend on both level
and spectrum. Some authorities also insist that the ear will tolerate
relatively high levels of jitter, up to a few nanoseconds, though that
has not been the experience of this magazine's writers. But as an
indicator of a product's ultimate resolution, these measurements of
jitter and noise floor are illuminating. The McIntosh, PrimaLuna, and
Krell are all disappointing in different ways, while the Simaudio,
Meridian, Bryston, Boulder, and Ayre measurements in particular indicate
the presence of some serious audio engineering talent on the staffs of
these companies. "

at - http://www.stereophile.com/content/case-jitters

Can you do similar tests with the RMAA suite ?

(BTW, what do you mean by - "Subjectively, likewise, the Touch sounds
great played through the Essence 1..." ? Also, "subjectively" do you
notice any difference in sound between the toslink and coax)


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[Julf]

SoftwireEngineer wrote:
> Anyways, this is what Stereophile has to say

"whatever you measure, you are still wrong, because we say so"? :)


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[Archimago]

SoftwireEngineer wrote:
> oh my goodness, I must be deluded :-) .. You guys need to come over to
> my place and blind test me :-)
> Anyways, this is what Stereophile has to say -
> ...
>
> at - http://www.stereophile.com/content/case-jitters
>
> Can you do similar tests with the RMAA suite ?
>
> (BTW, what do you mean by - "Subjectively, likewise, the Touch sounds
> great played through the Essence 1..." ? Also, "subjectively" do you
> notice any difference in sound between the toslink and coax)

Yes, I can do similar tests with the Dunn J-test like what Stereophile
talks about but obviously not at the level of resolution as the AP
machine John Atkinson utilizes. The test is different from the RightMark
data so far presented since RMAA cannot do the analysis.

That's why I asked earlier if anyone knows the FFT parameters / software
best used to analyze the data... I do agree with the quote when it said
papers have suggested the threshold of audibility is in the nanosecond
range since this has been my personal experience with simulated jitter
tests. These days, all competently engineered equipment should have
jitter measurements in the picosecond range which is likely an order of
magnitude below audibility.


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[Archimago]

ralphpnj wrote:
> Thanks again. Please understand that you are now damned for all eternity
> to the fires of hell, as are all audiophile heretics. Oh well, them's
> the breaks.

LOL :-)

I admit I have committed the sin of audiophile apostacy. As I type, I'm
sure the papers of excommunication are being delivered to my humble
abode...


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[ralphpnj]

Archimago wrote:
> LOL :-)
>
> I admit I have committed the sin of audiophile apostacy. As I type, I'm
> sure the papers of excommunication are being delivered to my humble
> abode...

The really sad part is that it was and still is so easy for you save
yourself. All you need to do is exactly what the "professionals" over at
the audio magazine do which is to present the measurements (just as you
have done) but then declare that in spite of the measurements the Touch
TT3.0 sounds so much better than the stock Touch. Be sure to include all
the usual audiophile buzz words, such as veils, windows, sound stage,
micro-dynamics, etc. and you will be forgiven. Repent sinner before it's
too late!!!

By the way I humbly apologize to all the real professionals out there
for referring to the clowns at the audio magazines as "professionals".
What a joke.


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[Mnyb]

ralphpnj wrote:
> The really sad part is that it was and still is so easy for you save
> yourself. All you need to do is exactly what the "professionals" over at
> the audio magazine do which is to present the measurements (just as you
> have done) but then declare that in spite of the measurements the Touch
> TT3.0 sounds so much better than the stock Touch. Be sure to include all
> the usual audiophile buzz words, such as veils, windows, sound stage,
> micro-dynamics, etc. and you will be forgiven. Repent sinner before it's
> too late!!!
>
> By the way I humbly apologize to all the real professionals out there
> for referring to the clowns at the audio magazines as "professionals".
> What a joke.

Professional is doing it for a living you don’t have to be good at it if
you can fool someone to pay .And professional in what discipline being
an audiophile magazine clown some of them are very good at that :) they
write the stuff their advertisers want to and what the readers want to
read .
It's a very deep hole.
They have managed to convince their readership on so many
pseudo-scientific things that you have to backtrack >30 year to begin to
put it right again if even possible .
Best prognosis, the current cult practitioners die of natural causes and
the hobby can be reinvented in a more sane regime as it once was by
younger people (discus practical speaker placements instead of quantum
purifiers).
The insane pricing is a sign of that they are selling to a diminishing
crowd of true believers...


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[Julf]

Mnyb wrote:
> They have managed to convince their readership on so many
> pseudo-scientific things that you have to backtrack >30 year to begin to
> put it right again if even possible .

To some degree they are just riding the trend of people giving up even
trying to understand technology, and relying on second-hand information
and superstition instead. As technology got more complex, it turned into
"magical black boxes". How many people understand what is going on under
the bonnet of their car these days?

> Best prognosis, the current cult practitioners die of natural causes and
> the hobby can be reinvented in a more sane regime as it once was by
> younger people (discus practical speaker placements instead of quantum
> purifiers).

Unfortunately there are far too many web sites and forums trying to
impart the audiophile belief system to a younger generation of computer
audio enthusiasts... :(


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[Archimago]

Julf wrote:
> To some degree they are just riding the trend of people giving up even
> trying to understand technology, and relying on second-hand information
> and superstition instead. As technology got more complex, it turned into
> "magical black boxes". How many people understand what is going on under
> the bonnet of their car these days?

Good points.

The problem with audio I think is that it is an ephemeral property.
Compared to our primary sensory modality of vision which we can hold
static like a scene on a nice 4K TV (I checked one out the other day!
Fantastic!) and slowly consider the detail/color/brightness, audio comes
and goes in an instant making it impossible to evaluate accurately other
than through objective measurements. To make matters "worse" for the
obsessive audiophile, music is highly emotionally laden which on the one
hand excites our passions about it and makes music a part of global
culture as humans but also makes it just so much harder to be
"objective" in evaluating just the sound itself - after all, who
evaluates their gear with just test tones separated from the music they
love or are familiar with?

These days, most "high end", well engineered products in the digital
front-end (and probably amps within the rated power output but this is
more complicated) perform beyond the perception of human hearing...
There's nothing left to evaluate perceptually which would differentiate
products. Nonetheless, manufacturers need to differentiate themselves,
and consumers need to convince themselves likewise they got their
money's worth! Factors like cost, weight, nostalgia, brand reputation,
etc. take on a greater role for evaluation yet of course no audiophile
reviewer would ever admit these peripheral factors are ever the primary
determinants of preference. Just have a read of the interviews with
Peter Qvortrup (Audio Note) to see how the man mixes the high-end
lifestyle with implications of sound quality based on bizarre
pseudoscience to tout his brand of technically obsolete equipment.

Oh a side note, I think other issues can creep into this as well. The
whole idea expounded by Neil Young about "data rate" = "sound quality"
is one of his justifications for going hi-res, for example. Personally,
I find this less morally objectionable (sure, to a point better data
rate for sampling helps) than the crazed "lifestyle of the rich and
famous" conspicuous consumption of Peter Q.


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[SoftwireEngineer]

I have to agree with Archimago on how difficult it is to evaluate audio
equipment. Music is continually varying and we do most of the
measurements with a single frequency which puts the system to a
"steady-state". It is quite possible the system will behave differently
with actual music. One of the reasons, we have measurements like Impulse
response, Damping factor, square wave output shape etc to evaluate how
the system would perform with real music. It is also quite possible we
have not come up with a more accurate/relevant measurements yet. This is
the reason I do give the benefit of doubt to audio reviewers. It is easy
to spot who is more reasonable if you follow their writings for a while.
For eg. recently reading about the NCORE amplifier module from Hypex,
one writer wrote that some companies are upset that Hypex founders have
started their own company to sell finished products which will put a
damper on their plans to sell the NCORE amps with fancy cases and high
price tags. Even though price does not go up linearly with performance,
many times, the pricing of high-end audiophile products are obnoxious.
Personally, for me, having born and grown up in a poor country (India)
and living half my life in the US, I still cannot get myself to spend so
much on audio (even though I do some charity to some extent) because of
this "obnoxious pricing".
Based on my following of these high-end companies - many just push the
specs wherever they can, even if they translate into improved sound or
not. Sometimes, I feel they themselves do not know why it sounds better
(I do know one designer mentioning this), if at all it sounds better.
Working on one aspect of the system but indirectly/unexpectedly resolve
some other issue. So we may have to give them some room for just
"trying". How much we are willing to pay for this is a different story
:-)

Right now, I am enjoying music on my Newform Research R630 with only one
woofer. I have to keep the speakers close the the front wall for some
bass reinforcement. Just to make a point about how speaker
placement/room acoustics is more important than many other things
electronic. IMHO, my speakers are not very good looking and the designer
is more reasonable and less "greedy" than some audio manufacturers, who
would otherwise put fancy bases, industrial design on these same
speakers, advertise a lot and sell for much more money.
One needs to hunt for value in this world.


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[Archimago]

SoftwireEngineer wrote:
>
> Based on my following of these high-end companies - many just push the
> specs wherever they can, even if they translate into improved sound or
> not. Sometimes, I feel they themselves do not know why it sounds better
> (I do know one designer mentioning this), if at all it sounds better.
> Working on one aspect of the system may indirectly/unexpectedly resolve
> some other issue. So we may have to give them some room for just
> "trying". How much we are willing to pay for this is a different story
> :-)
>
> Right now, I am enjoying music on my Newform Research R630 with only one
> woofer. I have to keep the speakers close the the front wall for some
> bass reinforcement. Just to make a point about how speaker
> placement/room acoustics is more important than many other things
> electronic. IMHO, my speakers are not very good looking and the designer
> is more reasonable and less "greedy" than some audio manufacturers, who
> would otherwise put fancy enclosures, industrial design on these same
> speakers, advertise a lot and sell for much more money.
> One needs to hunt for value in this world.

R630 - looks like a nice speaker system. One woofer? As in the other one
blew out or something!? Looks like you need to get it repaired STAT if
so!

As you alluded to, speakers and room interactions are absolutely
essential. Curious when you talk to the designers, are blind listening
test used as the ultimate arbiter of whether something sounds better or
would they use some other methodology?


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[SoftwireEngineer]

@Archimago - my speakers are fine. Just wanted to point out because of
less number of drivers (than my previous Silverline Sonatina) it needs a
different position.
I have seen speaker designers say they do not shoot for a flat frequency
response in an anechoic chamber, because with real music it does not
sound good. I am not sure they are just trying to pander to the
"audiophile" crowd with this statement. I personally, go for specs as
well as sound. Many audiophile brands are not true "high-fidelity"
systems. They are tuned to sound in a certain way. One of the Integrated
Amps I had earlier 'AudioRefinement Complete' (ralph has it, I think)
was found to be designed purposely with some non-linear compression by
hackers ie. louder treble sounds get attenuated more. Apparently, this
is done to voice it like tube amps. I have not liked this at all. The
Primare brand is also voiced similarly based on my listening at the
stores here. One of the reasons I like my true digital amplifiers (like
TACT and Panasonic) very transparent. An audio reviewer also mentions he
uses the current Nuforce D100 as the ultimate reference for
transparency.
I prefer 'live/dynamic/realistic/transparent' sound. My Newform seems a
little bit "polite" compared to Magnepans which was the speakers I
really liked for the 'live' sound.
Sound can be created in many ways (dynamic, planar/ribbon, etc and
solid-state/tube etc) it is difficult to say which one is really
high-fidelity, I guess (some people say, micro-dynamics are good with
tubes, I personally dont like that sound).


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[ralphpnj]

SoftwireEngineer wrote:
> @Archimago - my speakers are fine. Just wanted to point out because of
> less number of drivers (than my previous Silverline Sonatina) it needs a
> different position.
> I have seen speaker designers say they do not shoot for a flat frequency
> response in an anechoic chamber, because with real music it does not
> sound good. I am not sure they are just trying to pander to the
> "audiophile" crowd with this statement. I personally, go for specs as
> well as sound. Many audiophile brands are not true "high-fidelity"
> systems. They are tuned to sound in a certain way. One of the Integrated
> Amps I had earlier 'AudioRefinement Complete' (ralph has it, I think)
> was found to be designed purposely with some non-linear compression by
> hackers ie. louder treble sounds get attenuated more. Apparently, this
> is done to voice it like tube amps. I have not liked this at all. The
> Primare brand is also voiced similarly based on my listening at the
> stores here. One of the reasons I like my true digital amplifiers (like
> TACT and Panasonic) very transparent. An audio reviewer also mentions he
> uses the current Nuforce D100 as the ultimate reference for
> transparency.
> I prefer 'live/dynamic/realistic/transparent' sound. My Newform seems a
> little bit "polite" compared to Magnepans which was the speakers I
> really liked for the 'live' sound.
> Sound can be created in many ways (dynamic, planar/ribbon, etc and
> solid-state/tube etc) it is difficult to say which one is really
> high-fidelity, I guess (some people say, micro-dynamics are good with
> tubes, I personally dont like that sound).

Yes I do have an 'AudioRefinement Complete' integrated amp and I use in
my bedroom system where it drives a pair of Energy Veritas 2.0 bookshelf
speakers. I would not call this setup particularly high end but it does
sound quite nice. I've used other amps to drive the Energy speakers and
no other amp sounds as good as the AudioRefinement. For some reason
these two (three, really) pieces work very well together.

On your other points regarding the many different methods available to
reproduce recorded sounds and music, this is something that is often
overlooked by many people when they try to make definitive statements
about audio, such as "the Quads are the best speakers" or "Krell amps
are the best". For each different method there are a different set of
positives and negatives and not everyone agrees on which method sounds
best for them. So some people like dynamic speakers driven by solid
state class A/B amps while other people prefer point source speakers
driven by tube amps. To which I say, hey it's money so listen to
whatever sounds best to you.


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[Archimago]

Before packing up the Touch and moving it back upstairs to my bedroom
music system, I thought I'd try it in combination with the AUNE X1 which
I had previously used as an adaptive USB DAC (I see that newer models
have the Tenor 24/96 USB chips).

This is an example of the level of functioning that a $200 DAC buys you
mail order off eBay these days from Asia.

Result:
14398
Summarized above are the results with TosLink/Coax at 16/44, 24/96, and
24/192 (coaxial only, unlike the Essence One). Around -110dB noise floor
with 24-bit audio and -94dB with 16-bits. Note that this is with just
RCA output which is phenomenal!

Just a few graphs for completeness...
16/44 frequency response & THD:
1439914400

24/96 frequency response & THD:
1440114402

What does $200 buy you in terms of sound quality? Quite a lot!


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[Archimago]

ralphpnj wrote:
> Thank you Archimago. Truly amazing work. Very entertaining and highly
> enlightening. What I take away from all these test results is that
> digital audio is really quite different from analog audio and to keep
> insisting on applying the same belief system, e.g. heavy duty and
> expensive cables instead of well made less expensive cables, to digital
> audio is never going to improve the sound. Improving digital audio
> requires a different understanding and different ideas so the sooner the
> audiophile understands this the sooner improvements in digital audio
> will happen.
>
> Thanks again for helping to enlighten all of us.

Thanks Ralph.

I am looking forward to getting the gear together for the Transporter
test in the basement when I have time. I'll likely be doing the tests
with my laptop (on battery) and the power conditioned system downstairs
may give me even better results free from the (electrically) noisy Intel
i7 I've so far been using. Plus I'll get a chance to see what difference
balanced vs. unbalanced off the Transporter makes.


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[Archimago]

Alright, ran some tests on the Touch to see how well it works doing
transport duty!
14362


Setup:
Touch with EDO firmware off the app gallery installed. For the sake of
clarity, when I indicate a test was run in "EDO mode", this means it's
the digital only mode, otherwise I'm referring to the default "digital +
analogue" mode.

The Touch was connected by ethernet to the basement music server -->
either TosLink or coax out --> ASUS Xonar Essence 1 DAC --> XLR cables
--> E-MU 0404 USB for measurement.

As previously measured, the balanced XLR output from the Essence One has
so far given me the best electrical noise suppression and lowest noise
floor, the Touch was used as the digital transport for this system. I
then played the RightMark calibration, and test tones off the Touch and
measured the analogue out from the Essence 1.

About the digital cables:
TosLink - cheap "VITonet" labelled plastic fibre cable - no idea where I
got this, if I bought it, would likely be <$10 at local supplies store.
Pretty thin and flimsy looking but gives a good tight connection with
the ends.

Coaxial - this gave me an opportunity to try a simple unshielded 3'
stereo audio RCA cable (zip chord that I never used supplied with an old
DVD player! forget any impedance matching or electrical shielding) vs.
an actual shielded 6' coaxial cable "Acoustic Research Pro Series" I
bought 10 years ago for ~$20.

First off, 16/44:
14359

Nothing to see here! Essentially perfect measurements... The 1st column
is just the audio played to the Essence 1 through USB 2. Whether I used
TosLink, cheap RCA, actual coaxial to connect to the Essence 1 from the
Touch did not matter.

Now, 24/96 (standard "digital + analogue" mode):
14360

Hmmm, not unexpectedly, the "RCA as coaxial" is the stand out here.
Slightly reduced noise floor (~1 dB), reduced dynamic range (~2 dB),
mildly worse stereo crosstalk.

How about 24/96 in "EDO mode" with the analogue output turned off?
14361

Essentially the same as the standard mode. It appears that turning off
the analogue output circuitry (or at least silencing it) does not affect
the test results in any meaningful way. Again, the RCA cable performance
is inferior to a proper shielded coaxial cable. However one has to
realize that at this level of performance, the difference is really so
minor that it's unlikely anyone would be able to tell a difference from
listening! To give you an idea, here's the THD plot, notice how little
difference there is with the RCA cable (cyan) just rising over the
others like around the 5-10kHz range.

14363

Conclusion so far:
1. 16/44 performance is beyond reproach according to these tests.
2. 24/96 performance likewise is excellent. Starting to see the
limitations of an unshielded cable connected to the coaxial SPDIF but
even in such an extreme situation, the rise in noise floor likely
inaudible.

Later, I'll add the "EDO" 24/192 measurements!


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[Archimago]

In a couple week - once I finish up the results of the MP3 test, I'll
start cross-posting these measurements over to my blog with specific
pages for the components / tests. Wanted you guys to have a first look,
especially with the SB gear I have!

I must say, I've learned a lot about the limits of my gear and
maximizing quality of the output by doing this exercise... Some of the
troubleshooting when I got strange results have been particularly
enlightening about the thresholds of clipping and utilizing the E-Mu as
a recording device.

I'm in the process of putting together a "Touch as transport" series of
tests for later today as suggested by Mnyb. Just finishing up some
24/192 measurements with the EDO kernel... I'm actually very impressed
by the Touch's engineering and quality of components!


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[pippin]

Hi, let me chime in for one more question: did I overlook the test for
server-side FLAC decoding vs. FLAC decoding on the touch itself?
I would be really interested in that one, especially also in a
comparison between the WiFi vs. Ethernet performance.

The rationale is that counter to "conventional" audiophile wisdom my
experience when testing 96/24 playback in iPeng was that the raw PCM
data actually caused a noticeably _higher_ system load compared to FLAC.
What this means is, that he system load caused by decoding FLAC is
actually less than the load caused by the higher amount of data (roughly
factor 2) for PCM. It was quite noticeable even on a device as
performant as iPad 3.

Now, the iPad of course always uses WiFi (no Ethernet there) and it's
quite obvious that the load actually HAS to be higher: all the data
needs to go through the WPA decoding which certainly requires more power
than FLAC decoding.
But then I'm not sure that the WPA decoding isn't done in the WiFi
chipset so that the CPU (where I noticed the difference in load) doesn't
ever get involved.

Any chance you'd run one (OK, four) more test(s)?


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[Archimago]

pippin wrote:
> Hi, let me chime in for one more question: did I overlook the test for
> server-side FLAC decoding vs. FLAC decoding on the touch itself?
> I would be really interested in that one, especially also in a
> comparison between the WiFi vs. Ethernet performance.
>
> The rationale is that counter to "conventional" audiophile wisdom my
> experience when testing 96/24 playback in iPeng was that the raw PCM
> data actually caused a noticeably _higher_ system load compared to FLAC.
> What this means is, that he system load caused by decoding FLAC is
> actually less than the load caused by the higher amount of data (roughly
> factor 2) for PCM. It was quite noticeable even on a device as
> performant as iPad 3.
>
> Now, the iPad of course always uses WiFi (no Ethernet there) and it's
> quite obvious that the load actually HAS to be higher: all the data
> needs to go through the WPA decoding which certainly requires more power
> than FLAC decoding.
> But then I'm not sure that the WPA decoding isn't done in the WiFi
> chipset so that the CPU (where I noticed the difference in load) doesn't
> ever get involved.
>
> Any chance you'd run one (OK, four) more test(s)?

Hi Pippin; great work on iPeng BTW! Been using it for years :-)

On the 1st page of this thread for the Touch measurements "MEASUREMENT:
Logitech Squeezebox Touch (Pt 2)", I measured the WiFi vs. ethernet
conditions (16/44 & 24/96) using the Touch's analogue output and found
no difference. This was with "native" FLAC decoding on the device
itself. So you're interested in whether server side decoding (hence
higher data transmission of PCM) might add to the noise level of the
analogue outputs? Is that an accurate summary of what you'd like to see?


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[Wombat]

pippin wrote:
> The rationale is that counter to "conventional" audiophile wisdom my
> experience when testing 96/24 playback in iPeng was that the raw PCM
> data actually caused a noticeably _higher_ system load compared to FLAC.
> What this means is, that he system load caused by decoding FLAC is
> actually less than the load caused by the higher amount of data (roughly
> factor 2) for PCM. It was quite noticeable even on a device as
> performant as iPad 3.
>
> Now, the iPad of course always uses WiFi (no Ethernet there) and it's
> quite obvious that the load actually HAS to be higher: all the data
> needs to go through the WPA decoding which certainly requires more power
> than FLAC decoding.
> But then I'm not sure that the WPA decoding isn't done in the WiFi
> chipset so that the CPU (where I noticed the difference in load) doesn't
> ever get involved.
> Any chance you'd run one (OK, four) more test(s)?
I lately did read over hydrogen that one person noticed the same thing.
Seems he does program the audio decoding part of "Rockbox" and he
mentioned the higher load due to uncompressed formats against simply
using flac
WPA is decoded insided the WIFI card. I did read that in the specs of a
Philips WIFI card but canŽt find the document atm.

Besides that i have to thank you Achimago for making these experiments
and offering the rersults here!!
Now you only have to test audiophile CAT cable betweeen your router and
PC, otherwise all other benefits of mods of course canŽt come thru ;)


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[ralphpnj]

Wombat wrote:
> Besides that i have to thank you Achimago for making these experiments
> and offering the rersults here!!
> Now you only have to test audiophile CAT cable betweeen your router and
> PC, otherwise all other benefits of mods of course canŽt come thru ;)

And don't forget to use the new Audiophile Air Purifier and Ion
Alignment Filtering System in your room whenever you are using WiFi
since without properly filtering and treating the air the WiFi signal
will be adversely affected. Using the Audiophile Air Purifier and Ion
Alignment Filtering System will result in a NIGHT & DAY improvement in
the sound of all streamed audio. A veil will be lifted and a window
opened on the sound. In fact the Audiophile Air Purifier and Ion
Alignment Filtering System is nothing short of revolutionary. Read the
full review at ubeenhad.com.

Please note: The Audiophile Air Purifier and Ion Alignment Filtering
System is available at the introductory price of $5,000 for a limited
time only. Please PM me for full details.


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[Mnyb]

ralphpnj wrote:
> And don't forget to use the new Audiophile Air Purifier and Ion
> Alignment Filtering System in your room whenever you are using WiFi
> since without properly filtering and treating the air the WiFi signal
> will be adversely affected. Using the Audiophile Air Purifier and Ion
> Alignment Filtering System will result in a NIGHT & DAY improvement in
> the sound of all streamed audio. A veil will be lifted and a window
> opened on the sound. In fact the Audiophile Air Purifier and Ion
> Alignment Filtering System is nothing short of revolutionary. Read the
> full review at ubeenhad.com.
>
> Please note: The Audiophile Air Purifier and Ion Alignment Filtering
> System is available at the introductory price of $5,000 for a limited
> time only. Please PM me for full details.

I use only imported spring time air from Tibet , my sound waves should
not propagate thru that dross that can be found outside the door ;) or
sometimes air from the actual venue ( but the stale beer and smoke from
some jazz clubs gets to you ).


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[ralphpnj]

Mnyb wrote:
> I use only imported spring time air from Tibet , my sound waves should
> not propagate thru that dross that can be found outside the door ;) or
> sometimes air from the actual venue ( but the stale beer and smoke from
> some jazz clubs gets to you ).

Interesting idea using the air from the actual venue. I wonder how much
some wealthy audiophile would pay for the air from The Pawnshop (as in
"Jazz at the Pawnshop") captured during the time the actual recording
was made. I would guess that with a few liters of that rare air one
would be set for life :)

Anyway are you interesting in acquiring one of the first Audiophile Air
Purifier and Ion Alignment Filtering Systems or are you just wasting my
time? :)


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[Mnyb]

ralphpnj wrote:
>
>
> Anyway are you interesting in acquiring one of the first Audiophile Air
> Purifier and Ion Alignment Filtering Systems or are you just wasting my
> time? :)

ion alignment can I polarise all air molecules so that the radio waves
travels faster then ( how should I align the antenna ) ;)

But I heard that aligned ions interfere with my shakti hallographs .

I better load up some of the vintage Woodstock air , you get in a funny
mode for sure ;)


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[Archimago]

You guys are hilarious :-).

Well, it's cloudy and 6C today in Vancouver, humidity 80%, so probably a
terrible time to do any real audiophile listening! Guess I'll have to
wait a month or two before humidity gets to 60% so I can "seriously"
listen (or import that crisp Tibetan air). Sorry Ralph, your machine
will likely lead to parasitic vibrations with the Peter Belt foils I
have all over my DAC, cables, and CD's - just can't have that kind of
thing going on.

OK, I'll give the following test a try tonight:

Touch analogue out measurement (IMO no point with digital out since
measurements of noise floor, dynamic range, etc. are just showing the
limits of the downstream DAC as demonstrated by the lack of difference
shown despite how extreme everything went when I tried the TT mod!):

1. Server side decoding of FLAC --> PCM 24/96. Too bad 24/192 cannot be
played analogue on the Touch since that would REALLY stress the
wireless.
2. WiFi with WPA password protection in place, I'll try to report
wireless strength. Remember, the Touch is only capable of Wireless-G on
the 2.4GHz band. (My wireless router is an ASUS RT-N56U bought about 2
years ago if anyone cares.)
3. Hypothetically, then, if indeed this amount of data throughput is
stressing the Touch, then we would expect the noise level to rise due to
all that electrical "work" being done, right? Obviously, if the wireless
network cannot handle the throughput, the result should show gross
anomalies due to buffer issues as well.
4. Just to show that I'm not completely a measurebator, I'll subject
myself to listening to some "Waltz For Debbie" (24/192 -> 24/96 server
downsampling) and see if Bill Evans and boys sound like crap when the
music is "beamed".

Cheers... Let me know if there's anything else you want to see done.
After this, I really can't imagine anything else I care to test with the
Touch.


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[Mnyb]

Archimago wrote:
> You guys are hilarious :-).
>
> Well, it's cloudy and 6C today in Vancouver, humidity 80%, so probably a
> terrible time to do any real audiophile listening! Guess I'll have to
> wait a month or two before humidity gets to 60% so I can "seriously"
> listen (or import that crisp Tibetan air). Sorry Ralph, your machine
> will likely lead to parasitic vibrations with the Peter Belt foils I
> have all over my DAC, cables, and CD's - just can't have that kind of
> thing going on.
>
> OK, I'll give the following test a try tonight:
>
> Touch analogue out measurement (IMO no point with digital out since
> measurements of noise floor, dynamic range, etc. are just showing the
> limits of the downstream DAC as demonstrated by the lack of difference
> shown despite how extreme everything went when I tried the TT mod!):
>
> 1. Server side decoding of FLAC --> PCM 24/96. Too bad 24/192 cannot be
> played analogue on the Touch since that would REALLY stress the
> wireless.
> 2. WiFi with WPA password protection in place, I'll try to report
> wireless strength. Remember, the Touch is only capable of Wireless-G on
> the 2.4GHz band. (My wireless router is an ASUS RT-N56U bought about 2
> years ago if anyone cares.)
> 3. Hypothetically, then, if indeed this amount of data throughput is
> stressing the Touch, then we would expect the noise level to rise due to
> all that electrical "work" being done, right? Obviously, if the wireless
> network cannot handle the throughput, the result should show gross
> anomalies due to buffer issues as well.
> 4. Just to show that I'm not completely a measurebator, I'll subject
> myself to listening to some "Waltz For Debbie" (24/192 -> 24/96 server
> downsampling) and see if Bill Evans and boys sound like crap when the
> music is "beamed".
>
> Cheers... Let me know if there's any last thing you want to see done
> while I have the Touch connected.

Thanks for this measurements series a lot of real work done .

4. Urgh , very fine recording but you tikled another toe here , the
intrisinic resolution of the old recordings are less than any modern
format you chose to present them in . Even if it's fantastic sounding
with fantastic music the noise levels and frequency response are such
that it can be completely coded by any format from 16/44.1 and upward.
The delivery format is never the limit nowadays .


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[SoftwireEngineer]

Mnyb wrote:
> Thanks for this measurements series a lot of real work done .
>
> 4. Urgh , very fine recording but you tikled another toe here , the
> intrisinic resolution of the old recordings are less than any modern
> format you chose to present them in . Even if it's fantastic sounding
> with fantastic music the noise levels and frequency response are such
> that it can be completely coded by any format from 16/44.1 and upward.
> The delivery format is never the limit nowadays .

I have heard 16bit/44.1Khz that is as good as any other high-resolution.
I think it all depends on the mic'ing techniques and the equipment used
to record. To actually, compare low vs hi-res it is always good to
downsample the high-res to low-res and then compare (instead of using a
different low res recording). Usually, the high-res recordings are made
with more care, better equipment and that is where the improvement comes
from usually (my opinion). Quite likely beyond 24bit/48Khz there is any
improvement. If at all there is, it is probably due to implementation
reasons (the high-res bits taking a different path in the system than
the low res) vs actual improvement from real resolution changes.


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[Mnyb]

SoftwireEngineer wrote:
> I have heard 16bit/44.1Khz that is as good as any other high-resolution.
> I think it all depends on the mic'ing techniques and the equipment used
> to record. To actually, compare low vs hi-res it is always good to
> downsample the high-res to low-res and then compare (instead of using a
> different low res recording). Usually, the high-res recordings are made
> with more care, better equipment and that is where the improvement comes
> from usually (my opinion). Quite likely beyond 24bit/48Khz there is any
> improvement. If at all there is, it is probably due to implementation
> reasons (the high-res bits taking a different path in the system than
> the low res) vs actual improvement from real resolution changes.

Yep i've done a lot of that downsampled hirez ,but I've used modern
24/192 /96 recordings so that is reasonable possibility of it making a
difference ,sorry even if I'm a fanboy of the DVDA format ( have
hundreds ) no difference .

A good master is good master , the reality is that in some cases a much
better version/master can be sold under the hirez moniker.
This is apparently a needed marketing gimmick as the record companies
have destroyd " remastered " as a quality seal .
Nowadays "remastered" usually means crap , they tried to make it sound
modern with digital clipping and squashed dynamics etc the late 80's
early 90's CDs's of some stuff is usually the best bet .

Like a coincidence , if DVDA or SACD has a cd quality layer ,this layer
always sounds worse than the hirez ;)


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[SoftwireEngineer]

Mnyb wrote:
> Yep i've done a lot of that downsampled hirez ,but I've used modern
> 24/192 /96 recordings so that is reasonable possibility of it making a
> difference ,sorry even if I'm a fanboy of the DVDA format ( have
> hundreds ) no difference .
>
> A good master is good master , the reality is that in some cases a much
> better version/master can be sold under the hirez moniker.
> This is apparently a needed marketing gimmick as the record companies
> have destroyd " remastered " as a quality seal .
> Nowadays "remastered" usually means crap , they tried to make it sound
> modern with digital clipping and squashed dynamics etc the late 80's
> early 90's CDs's of some stuff is usually the best bet .
>
> Like a coincidence , if DVDA or SACD has a cd quality layer ,this layer
> always sounds worse than the hirez ;)

DO you think they purposely mess up the CD layer ? I found the SACD/CD
player sound almost the same in my Alison Kraus's New Favorite. But then
again, I am listening to 24bit/88.2kHz output of my Oppo when playing
SACDs. For some reason, I am not sure about SACD quality. It seems a
little "smoothed" out. In contrast PCM recordings seem more
'transparent/natural' to me. My Magnepan speakers crossover cannot
handle SACD output (the NewForms seem fine) and I hear some noise when
playing SACDs even after it has been downsampled to PCM by the OPPO. Did
not try the analog out of the Oppo to my digital amp when playing SACDs.
Anyways, I have only around < 20 or 30 SACD/DVDAs.


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[SoftwireEngineer]

BTW, it is good to buy DVD-As or high-res file just to get hold of good
recordings. This is good mainly for classical music (different
orchestras playing classics can be recorded with new/updated equipment).
But for mainstream (rock, jazz etc) you are stuck with the original
recording and get lucky when they release better transfers to digital
from analog.


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[ralphpnj]

SoftwireEngineer wrote:
> BTW, it is good to buy DVD-As or high-res file just to get hold of good
> recordings. This is good mainly for classical music (different
> orchestras playing classics can be recorded with new/updated equipment).
> But for mainstream (rock, jazz etc) you are stuck with the original
> recording and get lucky when they release better transfers to digital
> from analog.

Another thing to consider is buying the DVD if available of a live
concert when the DVD has PCM audio since the audio used on the DVD is
usually less compressed than the CD counterpart. Same thing applies for
buying the vinyl if available since the vinyl is usually made using a
less compressed master. Of course all of the above apply for popular,
i.e. rock, music releases since most other musical genres remain
unaffected by the Loudness War.


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[Mnyb]

SoftwireEngineer wrote:
> DO you think they purposely mess up the CD layer ? I found the SACD/CD
> player sound almost the same in my Alison Kraus's New Favorite. But then
> again, I am listening to 24bit/88.2kHz output of my Oppo when playing
> SACDs. For some reason, I am not sure about SACD technology/paradigm. It
> seems a little "smoothed" out. In contrast PCM recordings seem more
> 'transparent/natural' to me. My Magnepan speakers crossover cannot
> handle SACD output (the NewForms seem fine) and I hear some noise when
> playing SACDs even after it has been downsampled to PCM by the OPPO. Did
> not try the analog out of the Oppo to my digital amp when playing SACDs.
> Anyways, I have only around < 20 or 30 SACD/DVDAs.

No in some cases it is not the same master at al ,it can be the old cd
"remaster" .

The audiophile labels may or may not act differently but how do you know
, Actually I have some chesky that's sounds very similar so they probaly
have transcoded the same master to fit the CD in that case

What I,loved about DVDA was discrete multichannel ,that is awesome ,the
amount of information you get bombarded with is staggering . I even run
stereo recordings in " Trifield" to utilise my center channel .

But it is not practical for a lot people to have 5.1 playback at the
same quality level as their 2ch stereo , but IMHO,they could if they
where not chasing extremely expensive rainbows !
Like some cool aid drinkers using 30k$ DAC's and then some SET tube amps
? This performs like something's from Mediamarkt or Walmart ( for us
people ) but cost more than my complete small Meridian HT ( no I can't
afford the big ones I'm I have a normal jobb , no Ferrari on my drive
way )


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[Archimago]

Mnyb wrote:
> Thanks for this measurements series a lot of real work done .
>
> 4. Urgh , very fine recording but you tikled another toe here , the
> intrisinic resolution of the old recordings are less than any modern
> format you chose to present them in . Even if it's fantastic sounding
> with fantastic music the noise levels and frequency response are such
> that it can be completely coded by any format from 16/44.1 and upward.
> The delivery format is never the limit nowadays .

Yes, these old recordings don't need 24/192. However, I must say that
this particular recording - the HDTracks 2011 release seems to be a good
remaster with DR14 and no crazy hi-frequency hash.


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[ralphpnj]

Archimago wrote:
> Yes, these old recordings don't need 24/192. However, I must say that
> this particular recording - the HDTracks 2011 release seems to be a good
> remaster with DR14 and no crazy hi-frequency hash.

For anything recorded since the beginning of the stereo era (about the
mid 1950s) it's not the age of the recording but rather the quality of
the recording that matters. There are plenty of recordings from the
1950s and 1960s which sound as good as anything recorded today. For
example Miles Davis' "kind of Blue" is very well recorded and compares
favorably with most "modern" recordings.


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[Mnyb]

ralphpnj wrote:
> For anything recorded since the beginning of the stereo era (about the
> mid 1950s) it's not the age of the recording but rather the quality of
> the recording that matters. There are plenty of recordings from the
> 1950s and 1960s which sound as good as anything recorded today. For
> example Miles Davis' "kind of Blue" is very well recorded and compares
> favorably with most "modern" recordings.

Yes , i agree completely but that was not the kind of quality i meant ,
the quality of excellent studio work is everlasting :)

But rather if you teleported a modern equipment back in time and hid
inside the equipment they used , would you not get something with less
noise ( you can hear noise cant you ) and more extended frequency
response in both bass and treble ? And these recording has a certain
tone to them don't you think , this is the "flavour" of the vintage
equipment or you could call it distortion .

I'm after the technical quality of the raw signal something 30hz - 15kHz
-70dB s/n does not take 24/192 to encode ( 0Hz - 96kHz -144dB s/n) .

As discussed you hear the limitations of the equipment of the era in
early multi tracking and other sound layering and effect passes . What
happens is that one iteration trough one of these vintage tape machines
is ok but it quickly detoriates trough generational losses mostly added
noise , that tells you something of these machines .

And yes it is completely possible to have completely horrible recordings
with perfect " technical quality " :)
The studio work and artistry can be hideous but in perfect 24/192
quality .
I rather see "sound of music" or any other technicolor flick than an
IMAX rendering of the city dump even if one of them have higher
intrisinic technical quality ?

And they knew what they did back then they rather used the limitations
artistically tape (machine saturation ) etc and knew to make it work for
the music , you can hear on voice recordings that special tone ,probably
due the microphones used .

But these recordings sounded better than delivery format of the era the
LP .
What I've also learned is that they knew that the LP would never sound
the same as the master so they tweaked it a bit so when cutting it it
would come out more pleasant in the end product . Therefore a "LP master
tape" migth not be the best choice for remastering .


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[SoftwireEngineer]

Archimago wrote:
> Yes, these old recordings don't need 24/192. However, I must say that
> this particular recording - the HDTracks 2011 release seems to be a good
> remaster with DR14 and no crazy hi-frequency hash.
>
> I want to see if i get any snap/crackle/pop with 24/96 streamed
> uncompressed over WiFi.
Well.. it all depends on how strong your wifi connection is, right ? I
have only 30-40% and with 24/96 uncompressed I do get
interruptions/rebuffering pauses. Maybe, I will switch to compressed
(after I hear back from pippin).
@ralph - I have Davis's Kind of Blue. Agree with you. Very good
recording. Even in 50s/60s there was good analog recording
technologies/tape systems.


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[ralphpnj]

SoftwireEngineer wrote:
> Well.. it all depends on how strong your wifi connection is, right ? I
> have only 30-40% and with 24/96 uncompressed I do get
> interruptions/rebuffering pauses. Maybe, I will switch to compressed
> (after I hear back from pippin).
> @ralph - I have Davis's Kind of Blue. Agree with you. Very good
> recording. Even in 50s/60s there was good analog recording
> technologies/tape systems.

And don't forget that until the mid-1960s multi-track recording was not
very common, if in use at all, so almost all recordings were done live
in the studio with everyone playing together. Which is one reason why so
many modern recordings sound as if they were recorded at twenty
different times in twenty different places - because they were! I'll
take live in the studio, particularly for jazz, anyday instead of an
over dubbed multi-tracked recording.


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[Archimago]

Okay, got home and did some testing with the Touch and 24/96 with LMS in
my basement PC doing server-side decompression of FLAC (labelled as "PCM
FLAC" in my chart below), streaming uncompressed PCM/WAV over to the
Touch.

Setup:
Touch stock firmware --> RCA outs (standard shielded stereo cable) -->
E-Mu 0404USB

Results:
14412

As indicated in the chart, for WiFi, I get signal strength of ~70% here.
The numbers are all pretty consistent. One interesting hint of
"something" might be the slightly higher stereo crosstalk in the "PCM
FLAC" tests (with both ethernet and WiFi) but this is generally around
+0.5dB and usually within the inter-test variability; however, I ran the
tests 3 times in each of these conditions and I think this slight
difference could be real (though obviously unlikely to be audible!).

Conclusion:
There you have it; I don't think there's an audible issue whether you
stream FLAC or PCM at least with test signals and without buffering
issues. One thing I notice is that with WiFi, there's a bit more latency
to the interface on the Touch... Just not as snappy switching tracks or
showing the album cover image. Maybe this effect biases one's
experience negatively?

"Waltz For Debby" sounds great so far without buffering issues ~30
minutes now.

Hopefully I get to my Transporter this weekend :-)


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[Mnyb]

Archimago wrote:
> Okay, got home and did some testing with the Touch and 24/96 with &
> without LMS in my basement PC doing server-side decompression of FLAC
> (labelled as "PCM FLAC" in my chart below), streaming uncompressed
> PCM/WAV over to the Touch. WiFi is 802.11g (2.4GHz) WPA password
> protected, ASUS RT-N56U router. The ethernet is a DLink gigabit switch
> (sorry, someone stole my $500/m "audiophile" CAT6, so making do with a
> generic white CAT5).
>
> Setup:
> Touch stock firmware --> RCA outs (standard shielded stereo cable) -->
> E-Mu 0404USB
>
> Results:
> 14412
>
> As indicated in the chart, for WiFi, I get signal strength of ~70% here.
> The numbers are all pretty consistent. One interesting hint of
> "something" might be the slightly higher stereo crosstalk in the "PCM
> FLAC" tests (with both ethernet and WiFi) but this is generally around
> +0.5dB and usually within the inter-test variability; however, I ran the
> tests 3 times in each of these conditions and I think this slight
> difference could be real (though obviously unlikely to be audible!).
> Evidence that having more data be processed by the network controller
> has a mild effect?
>
> Conclusion:
> There you have it; I don't think there's an audible issue whether you
> stream 24/96 FLAC or PCM at least with these test signals and without
> buffering issues. One thing I notice is that with WiFi, there's a bit
> more latency to the interface on the Touch... Just not as snappy
> switching tracks or showing the album cover image. Maybe this effect
> biases one's experience negatively?
>
> The other implication is that the Touch DAC is well shielded from the
> wireless electronics to not affect the inherent noise floor. As I said
> before, I think the Touch was engineered very well for such a small and
> inexpensive unit!
>
> "Waltz For Debby" sounds great so far without buffering issues ~30
> minutes now server-side decompressed at 24/96 over WiFi.
>
> Hopefully I get to my Transporter this weekend :-)

Great ! It also seams like Wifi has a bit higher distortion a whooping
0.0002 % more :) atrocious .

So the FLAC/WAV thing is also an Nigerian letter to the audiophiles .
You can say that flac is "better" on a Touch but that would be silly
with the very small differences .
The most interesting thing is that the ones claiming that they hear this
difference often uses external DAC's too ?


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[Archimago]

Mnyb wrote:
> Great ! It also seams like flac has a bit higher distortion a whooping
> 0.0002 % more :) atrocious .
>
> So the FLAC/WAV thing is also an Nigerian letter to the audiophiles .
> The most interesting thing is that the ones claiming that they hear this
> difference often uses external DAC's too ? Where the difference would be
> non existing .

:-)

Important to keep in mind that these are measurements with test tones
which of course places a different kind of load on the electronics. Real
music with all its intricacies may of course lead to a different kind of
response.


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[Jeff52]

Mnyb wrote:
> Like a coincidence , if DVDA or SACD has a cd quality layer ,this layer
> always sounds worse than the hirez ;)

Check out the CD layers of the remastered Rolling Stones and Bob Dylan
hybrid SACDs, in particular, Let It Bleed by the Stones. The CD layers
benefit from the DSD remastering of the SACD layer and really sound
good.


ralphpnj wrote:
> For anything recorded since the beginning of the stereo era (about the
> mid 1950s) it's not the age of the recording but rather the quality of
> the recording that matters. There are plenty of recordings from the
> 1950s and 1960s which sound as good as anything recorded today. For
> example Miles Davis' "kind of Blue" is very well recorded and compares
> favorably with most "modern" recordings.

I quite agree about Kind Of Blue. Some other good ones: Getz/Gilberto
SACD (2011 Analogue Productions remaster), Getz & Byrd Jazz Samba (1994
DCC remaster). Of course they would sound better with a Shunyata
Research ZTRON Anaconda power cable, but unfortunately I'm stuck with
the stock power cord for my Bryston amp.


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[Mnyb]

Jeff52 wrote:
> Check out the CD layers of the remastered Rolling Stones and Bob Dylan
> hybrid SACDs, in particular, Let It Bleed by the Stones. The CD layers
> benefit from the DSD remastering of the SACD layer and really sound
> good.
>
>
>
>
> I quite agree about Kind Of Blue. Some other good ones: Getz/Gilberto
> SACD (2011 Analogue Productions remaster), Getz & Byrd Jazz Samba (1994
> DCC remaster). Of course they would sound better with a Shunyata
> Research ZTRON Anaconda power cable, but unfortunately I'm stuck with
> the stock power cord for my Bryston amp.

Any Dylan in particular ( I find stones overrated but a good live show )
I sometime listen to Dylan but not often .

If I charge my phone via the Shunyata research would my mp3's sound
better ? I don't find the background black I'm rather surrounded by
busses and bicycles and people .
( this idea is not more crazy than thinking filling a squeezebox buffer
with a server with a tweaked OS or some such sounds better than data
otherwise gathered , I get " better " electrons in my iPhone ).


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[Jeff52]

Mnyb wrote:
> Any Dylan in particular ( I find stones overrated but a good live show )
> I sometime listen to Dylan but not often .
>
> If I charge my phone via the Shunyata research would my mp3's sound
> better ? I don't find the background black I'm rather surrounded by
> busses and bicycles and people .
> ( this idea is not more crazy than thinking filling a squeezebox buffer
> with a server with a tweaked OS or some such sounds better than data
> otherwise gathered , I get " better " electrons in my iPhone ).

The "Bob Dylan" album was not part of the SACD remasters but was
remastered in 16/44.1 in 2005. This version beats anything previously
released IMHO. I like all of the hybrid SACD remasters. There is an
excellent discussion here:
http://bobdylanisis.com/contents/en-uk/d94.html I'm a big fan of the
early Stones material and was very glad ABKCO decided to finally do
something about the lousy original recordings when they remastered the
albums. Considering the sources available, they did a great job I think.


The Shunyata needs to be connected to something actively playing music.
You will be unable to discern a difference if used merely to recharge
your iPhone. It will make mp3's sound like 24/192, with increased air
around the instruments and a major improvement in PRaT.


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[Julf]

Mnyb wrote:
> If I charge my phone via the Shunyata research would my mp3's sound
> better?

Of course. The electric field compensation circuit ensures the bits
settle in an unbiased state inside your phone - very important
considering all the eddy currents within the phone.

I wonder if there is a subtle message in the fact that they name their
products after snakes... :)


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[SoftwireEngineer]

Gentlemen, I know you like to have a laugh at the audiophile madness,
just like I am aghast at the obscene pricing and unnecessary cosmetic
build materials. In the case, of speaker cables and power cables, if you
measure the electrical properties of stock vs say, Shunyata, there will
surely be some difference. But does it translate to difference in the
sound of your system ? Probably not that much. To be fair, some
audiophile reviewers, they do say it does not benefit some very high end
equipment. So for power cables, it depends on the power supply in your
equipment. In the case of speaker cables, it all depends on the output
impedence of your amplifier and the impedence characteristics of your
speaker. All the speaker cable designers do, is keep tweaking the
inductance/capacitance balance. Many try to reduce the capacitance,
while keeping the inductance low (by winding which actually will
increase capacitance). Note, resistor, capacitor or inductor that we
buy/use in our circuits are not necessarily purely resistance,
capacitance or inductance. They have other properties as well, but very
low. It is quite possible some people can hear the differences between
different resistors, capacitors or inductors of the same value but from
different brand/quality. This is the same story with power or speaker
cables, they are not purely resistive.
BTW, Shunyata in sanskrit language means, nothingness. They are trying
to shoot for it in their cable designs.


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[ralphpnj]

SoftwireEngineer wrote:
> BTW, Shunyata in sanskrit language means, nothingness. They are trying
> to shoot for it in their cable designs.

I'm sorry to have to correct you on this but the nothingness has nothing
to do with their cable designs and everything to do with the contents of
your wallet, has in Shunyata is trying to achieve the nothingness state
for your wallet. And based on their prices I would say that they are
well on their to achieving their objective :)


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[Archimago]

FWIW since folks are talking power cables now, I compared the Essence
One's output using the stock power cable vs. hospital/biomedical power
cable used in ICU equipment. Obviously the biomed cable is much better
built and more robust, with better shielding.

No difference with the noise floor (around -113 dB). Remember, this is
all plugged into a relatively generic Belkin powerstrip with a noisy i7
nearby...


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[jvanhambelgium]

Archimago wrote:
> FWIW since folks are talking power cables now, I compared the Essence
> One's output using the stock power cable vs. hospital/biomedical power
> cable used in ICU equipment. Obviously the biomed cable is much better
> built and more robust, with better shielding.
>
> No difference with the noise floor (around -113 dB). Remember, this is
> all plugged into a relatively generic Belkin powerstrip with a noisy i7
> nearby...
>
> I can imagine a situation where a psychiatrist gets consulted to the ICU
> with the reason for the request being: "Middle aged post-MI 'audiophile'
> refuses repeat ECG unless done with Shunyata power cables." =-O


Did you perform some measurement between "regular" PSU of the Touch and
some ultra-lineair one ?
That would be interesting too.


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[Archimago]

jvanhambelgium wrote:
> Did you perform some measurement between "regular" PSU of the Touch and
> some ultra-lineair one ?
> That would be interesting too.

No. Don't have an (ultra)linear PS to test.


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[Julf]

SoftwireEngineer wrote:
> In the case, of speaker cables and power cables, if you measure the
> electrical properties of stock vs say, Shunyata, there will surely be
> some difference.

Right. Measureable electrical differences are one thing, and I would not
have anything against cable suppliers if they explained the properties
of their cables in those terms. Unfortunately they tend to resort to
snake oil pseudo-science gobbledegook such as "electric field
compensation circuits".


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[Mnyb]

Julf wrote:
> Right. Measureable electrical differences are one thing, and I would not
> have anything against cable suppliers if they explained the properties
> of their cables in those terms. Unfortunately they tend to resort to
> snake oil pseudo-science gobbledegook such as "electric field
> compensation circuits".

But you sell more :)

If you instead go like we made good speaker wire by having sufficiently
low R and low L and did this with astonishing technique of using more
copper and twist it ,now you pay 5000$ for it ? you get the 10$ it
really worh from me...

And power wires are totally bogus change the house wiring and the feeder
cable to the house then it might contribute , but no good product will
need this minuscule enhancent anyway ,if the voltage does not drop you
get more power probably fractions of dB :)

The mark-up on these snake cables must be astonishing if the
manufacturing cost exceeds 1% if their price they should reassess how
they do them so that they can make even more money , they dont actually
have to do all the gobbledegook they claim they are doing in their white
paper .
Who ever is going to do a metallurgic analysis on the cable or even cut
it open


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[Archimago]

It's Transporter time!

In order to do the measurements, I brought the gear downstairs to the
basement which is an electrically quieter environment. There's a Belkin
PureAV PF60 power center there for all the equipment. Also, the
measuring computer is now an AMD Phenom X4 laptop with Win 8 usually
used by my kids :-). The laptop was running on battery - I could detect
a 0.5dB difference with the AC adaptor plugged down at the -110dB noise
level.

One note about the XLR measurements you'll notice - the THD levels are a
bit higher than RCA. I believe this is a result of the fact that the
E-Mu 0404USB could not handle the XLR voltage from the Transporter and I
had to use the analogue attenuators (just barely!) avoid clipping.

Lets start with the 44kHz signal:
14421

Undoubtedly, the XLR output is significantly better than RCA. One
observation is that through the RCA's, the stereo crosstalk remained
around -90dB whether the signal was 16 or 24-bits. I'm not sure if this
is the limit of the Transporter itself (even the humble AUNE X1 could do
a measured -95dB), or has to do with the cables I used - a pair of
AudioQuest 6' interconnects. I don't remember which model of AudioQuests
these were (bought a few years ago in a moment of weakness :-) but they
are longer than the 3' I had been using to measure upstairs.

Now for 24/96:
14423

Again, we see the -90dB stereo crosstalk limit with the RCA output.
XLR's measure is fantastic! Likely hitting the performance limits of the
E-MU 0404USB and ~3dB better than the Essence One (of course the E1 did
not have the benefits of a power filter or low noise environment of the
laptop running on battery).

The WiFi router was in the same room as the Transporter hence the 90+%
wireless strength. No difference in the measurements whether WiFi or
ethernet.

To show in graph form the difference in noise floor between the RCA and
XLR:
14422
Notice the noise spikes like at 60Hz using the RCA cable (the AQ
construction seems to be shielded but can't confirm unless I cut it
open!).

Conclusion:
1. Overall the Transporter measures very well! Phenomenal XLR
performance - best I have been able to measure so far.
2. A bit unclear about that stereo crosstalk measurement with RCA
however. Seems a bit higher than I'd have expected. Might need to try a
different cable and see... (In case anyone wondering, I have not opened
up the Transporter to change the unbalanced volume attenuation.)


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[Archimago]

One more thing before leave the Transporter alone... I wanted to see if
turning on the TosLink effects loop affected measurements. Normally, I
have the Transporter --> TosLink --> Behringer DEQ2496 (room EQ) -->
TosLink --> Transporter as DAC, so it'd be nice to know that the DEQ in
digital mode doesn't affect the final sound.

With the DEQ2496 on bypass mode (ie. no room EQ processing):
14427

Using the Tributaries RCA cable as output, no difference whether the
DEQ2496 was digitally in line or not. Alas, by this time I had
disconnected the XLR's to double check and I didn't care to disrupt the
Transporter setup again :-)


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[garym]

Nice! Thanks for doing the Transporter analysis. I like (and use) the
XLR connections of my transporter.


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[SoftwireEngineer]

nordost and few others provide the measurements. Nordost's construction
is actually pretty fascinating.
It is cheaper to buy good copper or silver wire and Teflon and make it
yourself.


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[ralphpnj]

Archimago wrote:
> You guys are hilarious :-).

Thanks! And thanks once again for doing and reporting on all these
insightful tests.

I do have one thing to add regarding my wonderful air purifying machine
idea, which is why this idea rightful seen as crazy and comical but the
idea of audiophile grade CAT5/6 cable not also seen as crazy and
comical? And more importantly why don't the so called "professional" and
"knowledgeable" writers and editors in the audiophile press dismiss
these audiophile grade CAT5/6 cables as a complete waste of money as
they dismiss my wonderful Audiophile Air Purifier and Ion Alignment
Filtering System? (Hint the answer has to do with money.)


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[SoftwireEngineer]

pippin wrote:
> Hi, let me chime in for one more question: did I overlook the test for
> server-side FLAC decoding vs. FLAC decoding on the touch itself?
> I would be really interested in that one, especially also in a
> comparison between the WiFi vs. Ethernet performance.
>
> The rationale is that counter to "conventional" audiophile wisdom my
> experience when testing 96/24 playback in iPeng was that the raw PCM
> data actually caused a noticeably _higher_ system load compared to FLAC.
> What this means is, that he system load caused by decoding FLAC is
> actually less than the load caused by the higher amount of data (roughly
> factor 2) for PCM. It was quite noticeable even on a device as
> performant as iPad 3.
>
> Now, the iPad of course always uses WiFi (no Ethernet there) and it's
> quite obvious that the load actually HAS to be higher: all the data
> needs to go through the WPA decoding which certainly requires more power
> than FLAC decoding.
> But then I'm not sure that the WPA decoding isn't done in the WiFi
> chipset so that the CPU (where I noticed the difference in load) doesn't
> ever get involved.
>
> Any chance you'd run one (OK, four) more test(s)?

The higher system load is on ipads right ? Have you checked this out on
the Squeezebox itself (which is where this suggestion originated) ? This
probably is related to hardware architecture.


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[pippin]

Archimago wrote:
> Hi Pippin; great work on iPeng BTW! Been using it for years :-)
>
> On the 1st page of this thread for the Touch measurements "MEASUREMENT:
> Logitech Squeezebox Touch (Pt 2)", I measured the WiFi vs. ethernet
> conditions (16/44 &amp; 24/96) using the Touch's analogue output and
> found no difference. This was with "native" FLAC decoding on the device
> itself. So you're interested in whether server side decoding (hence
> higher data transmission of PCM) might add to the noise level of the
> analogue outputs? Is that an accurate summary of what you'd like to see?

Yes.

And thanks for the warm words on iPeng, good to hear you like it


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[pippin]

SoftwireEngineer wrote:
> The higher system load is on ipads right ? Have you checked this out on
> the Squeezebox itself (which is where this suggestion originated) ? This
> probably is related to hardware architecture.

It was on iPad, yes.
But the system load argument has also been brought forward for iPeng
where I can only observe that the load with PCM is higher.


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[AlexM]

About a year ago I replied to the TT3 thread with some measurements
comparing CPU utilisation in the Touch at 16/44 and 24/96 with and
without server-side FLAC decoding. For my tests, I used a wired Ethernet
connection. CPU utilisation was measured simply by connecting via SSH
and running 'top'

I don't have the exact figures to hand, but IIRC at 16/44 the difference
was something like 10% (via FLAC) vs. 30% (PCM to SBT), and at 24/96 it
was (55-65% vs >90%). I couldn't detect any beneficial effect of
streaming PCM, but this did establish that the theory supporting
improved SQ with at least this component of TT3 mods was a bust.
Unsuprisingly nobody could think of a reason why more CPU utilisation
would be better, especially since the theory was based on the opposite
being true. Suprising that nobody bothered to check before, especially
as it was so easy to do.

Ah.. here we go.. found my tests on a different board -
http://theartofsound.net/forum/showthread.php?t=15594&highlight=Touch+CPU&page=8

Archimango - excellent work on the tests BTW. I also have a Emu 0404
USB, and have long been telling people what a cracking interface it is,
especially for the money!. Your figures do show it to be a very
transparent device for A/D. You have inspired me to do some testing of
my own to optimise the noise performance of my system.

Cheers,
Alex


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[Archimago]

AlexM wrote:
> About a year ago I replied to the TT3 thread with some measurements
> comparing CPU utilisation in the Touch at 16/44 and 24/96 with and
> without server-side FLAC decoding. For my tests, I used a wired Ethernet
> connection. CPU utilisation was measured simply by connecting via SSH
> and running 'top'
>
> I don't have the exact figures to hand, but IIRC at 16/44 the difference
> was something like 10% (via FLAC) vs. 30% (PCM to SBT), and at 24/96 it
> was (55-65% vs >90%). I couldn't detect any beneficial effect of
> streaming PCM, but this did establish that the theory supporting
> improved SQ with at least this component of TT3 mods was a bust.
> Unsuprisingly nobody could think of a reason why more CPU utilisation
> would be better, especially since the theory was based on the opposite
> being true. Suprising that nobody bothered to check before, especially
> as it was so easy to do.
>
> Ah.. here we go.. found my tests on a different board -
> http://theartofsound.net/forum/showthread.php?t=15594&highlight=Touch+CPU&page=8
>
> Archimango - excellent work on the tests BTW. I also have a Emu 0404
> USB, and have long been telling people what a cracking interface it is,
> especially for the money!. Your figures do show it to be a very
> transparent device for A/D. You have inspired me to do some testing of
> my own to optimise the noise performance of my system.
>
> Cheers,
> Alex

Thanks for the note Alex! Glad to see another 0404USB user :-). Let me
know if in your testing you find a good way to measure JITTER!

Good point about checking on the Touch's CPU utilization. This AM I
logged into the Touch and ran 'top -d1' to check CPU usage every second
while playing music. I chose Prince's 'Batdance' since I had a good
24/96 vinyl rip as well as the original 16/44; I figure a dynamic track
like this could give the Touch's FLAC decoding a nice work out. This is
with my Touch running default EDO firmware settings, ethernet
connected.

For MOST of the time, the CPU use looks like this:
24/96:
FLAC level 8 compression: 20-30% default view, 70-85% peak with spectrum
analyzer

16/44:
FLAC level 8 compression: 15-25% default view, 65-75% peak with spectrum
analyzer

Clearly turning on the spectrum analyzer "Now Playing" view eats up CPU
cycles significantly more than just the playback itself from what I
see.

About a couple of times over the 4 minute song, the jive_alsa process
wakes up in a big way and this pushes CPU use to 90-100% for about 5-10
seconds. I believe jive_alsa runs at realtime priority level and there's
a discussion about it elsewhere. Whether this changes sound quality, I
dunno.

SO even though I couldn't find any sonic change going the way of TT3
with turning off the screen and everything, I wonder if doing the
opposite - stressing out the CPU with 24/96 FLAC + spectrum analyzer
view running during playback lead to stuff like increasing noise
floor... Maybe I'll give this a try tonight.


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[SoftwireEngineer]

Looks like I also did the same experiments - measuring CPU using top.
Conclusion - system is less idle when you are you streaming PCM. The
sirq component is high, because, this is deferred processing of the
incoming network data (which is more in case of PCM). The system starts
a bit idle but as it goes into the song the sirq starts piling up and
then again the last 1 minute of the song system frees up.
If at all there is any sonic difference, it is not coming from a less
idle CPU. BTW, I dont think I am noticing any difference. Also, I am
able to stream 24/96 when I am streaming flac. So I am going to set it
to flac streaming and run with it for a while and later switch see any
subjective differences exist.


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[AlexM]

Archimago wrote:
> Thanks for the note Alex! Glad to see another 0404USB user :-). Let me
> know if in your testing you find a good way to measure JITTER!
>
> Good point about checking on the Touch's CPU utilization. This AM I
> logged into the Touch and ran 'top -d1' to check CPU usage every second
> while playing music. I chose Prince's 'Batdance' since I had a good
> 24/96 vinyl rip as well as the original 16/44; I figure a dynamic track
> like this could give the Touch's FLAC decoding a nice work out. This is
> with my Touch running default EDO firmware settings, ethernet
> connected.
>
> For MOST of the time, the CPU use looks like this:
> 24/96:
> FLAC level 8 compression: 20-30% default view, 70-85% peak with spectrum
> analyzer
>
> 16/44:
> FLAC level 8 compression: 15-25% default view, 65-75% peak with spectrum
> analyzer
>
> About a couple of times over the 4 minute song, the jive_alsa process
> wakes up in a big way and this pushes CPU use to 90-100% for about 5-10
> seconds. I believe jive_alsa runs at realtime priority level and there's
> a discussion about it elsewhere. Whether this changes sound quality, I
> dunno.
>
> SO even though I couldn't find any sonic change going the way of TT3
> with turning off the screen and everything, I wonder if doing the
> opposite - stressing out the CPU with 24/96 FLAC + spectrum analyzer
> view running during playback lead to stuff like increasing noise
> floor... Maybe I'll give this a try tonight.

Hi again,

Thanks again for doing these measurements - nice to have some data to
discuss rather than the usual subjective opinions for FLAC vs. WAV. It
would be nice to do some Null measurement - can you do this with
Rightmark?.

FYI, I use Triode's EDO plug-in, and a Cambridge Audio Azur 851C
upsampler as a DAC. It would be great to know if you can measure any
difference in noise or distortion in this worst-case scenario, but my
money is on it not making a measurable difference.

With this configuration, CPU utilisation stays at 35-40% with FLAC
playing at 24/192 (2L sampler). I usually stay with the 'now playing'
screen saver, but do have somthing of a weak spot for the old VU
meters!.

Cheers,
Alex


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[SoftwireEngineer]

@Alex - I dont think you need to worry about Jitter from the SBT. Your
851c has heavy buffering and processing which should take care of the
jitter. In my case, I goes to my TACT Direct Digital Integrated
amplifier, so I am always concerned about source jitter. I am not sure
how much processing/buffering my TACT S2150 has.


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[Archimago]

Lets see if the RightMark results deteriorate with increased CPU usage
(all done with WiFi @ 60% strength, EDO kernel, analogue output
measured):
14431

Far left, Touch playing at 24/96 SERVER SIDE DECODING, default "Now
Playing".

2nd left, Touch playing 24/96 SERVER SIDE DECODING, with Spectrum
Analyzer "Now Playing".

3rd left, Touch playing 24/96 SERVER SIDE DECODING, Spectrum Analyzer +
5 TTY sessions running "nice -n -20 top -d1".
- This means that there are 5 PuTTY sessions open, each one running
'top' at a high priority level and refreshing system status every
second. With server side decoding of the FLAC, this is the limit for me
at WiFi 60%. If I open another PuTTY, this results in constant
rebuffering errors (all the PuTTY's needing to send out a screen of data
every second).

4 & 5th left, Touch playing 24/96 NATIVE FLAC, Spectrum Analyzer + 5 or
10 TTY sessions with "nice -n -20 top -d1".
- With native FLAC decode, the WiFi has more room to "breathe" due to
data compression and can handle more TTY sessions without buffering
errors!

As you can see, the measured results are rock solid even with an
unreasonable number of TTY sessions open (idle CPU usage went from ~1-5%
to 30% with all these PuTTY sessions). What this tells me is the Linux
realtime scheduler in the Touch is robust. For example, lower priority
processes like the Spectrum Analyzer refreshes slower as the number of
TTY/top sessions increases and the GUI responsiveness goes down as well
but the sound quality remains unchanged as far as I can tell with
measurements.

Here's what my screen looked like with all these TTY sessions all logged
into the Touch:
14432

As for 24/192, via ethernet I just listened to Cat Steven's "Tea For The
Tillerman" (HDTracks), and now starting Neil Young's "Harvest" (DVD-A
rip) with 10 TTY sessions in the background running the high-priority
'top' with Spectrum Analyzer display. No audible problems with the Touch
connected to the Essence One by TosLink. I didn't bother measuring this
since it sounds great and I didn't feel like dragging out the XLR cables
again ;-)!

Bottom line: I still see no evidence of high CPU use causing significant
change to the audio output... If there is something, it must be below
the noise floor of the Touch DAC in any case. Likewise, the Touch seems
to have good headroom for 24/192 playback as well. IMO, it's more
important then to make sure there's enough data bandwidth (ie. use
ethernet when you can!) than worry about CPU usage causing audio
degradation.

Usual disclaimer of course is that the measurements are with test
signals which may not represent the same performance with real music.


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[Mnyb]

Wonderful measurements .

There is a tool called audio diffmaker , that you probably can use with
the EMU .
It would make it possible to use music , you could for example record a
complete song with one setup then do it again with another setup and
cancel them out and se what's left , you even listen to the "null"
signall.
I would also run several runs without change with diffmaker to get a
grip on the general randomness of the system.

Note that I do not think test signals are invalid , the Touch does not
know music and it would not suddenly change its transfer function due to
that it perceived that the data comes from music :)

Using music would serve two purposes taking care of the unknown that if
it really is something that did not show up measuring thd or im it will
be seen now in the residue signal.
And it would be harder to deny the results for a sceptic as you have
taking in the totality of the reproduction of the signal.

http://www.libinst.com/Audio%20DiffMaker.htm

What I see it can be a bit tricky to,get going with some interfaces
depending how they work the idea is to time align signals and then
subtract .


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[AlexM]

Archimago wrote:
> Lets see if the RightMark results deteriorate with increased CPU usage
> (all done with WiFi @ 60% strength, EDO kernel, analogue output
> measured):
> 14431
>
> Bottom line: I still see no evidence of high CPU use causing significant
> change to the audio output... If there is something, it must be below
> the noise floor of the Touch DAC. Likewise, the Touch seems to have
> decent headroom for 24/192 playback as well. IMO, it's more important
> then to make sure there's enough data bandwidth (ie. use ethernet when
> you can!) than worry about CPU usage causing audio degradation in
> standard usage. Server side decoding understandably increases the risk
> of needing to rebuffer and offers no advantage as far as I can tell
> compared to native FLAC decoding.
>
> Usual disclaimer of course is that the measurements are with test
> signals which may not represent the same performance with real music.

Thanks Archimango - good stuff. Nice to know that the analogue outs
aren't measurably affected by CPU utilisation. That is what my ears are
telling me too.

As a side issue, I notice that SP/DIF input to my 851c sounds quite a
bit better than via USB 2.0 High speed (or USB 1.0 Full speed). It
sounds quite soft and rolled off. I might try to replicate your tests
(seeing as I have the same gear) and see if this is measurable.

Cheers,
A.


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[SoftwireEngineer]

@Archimago - this is awesome. But it feels like work though (I was
benchmarking some Big Data code yesterday) :-)
If you want to just load the CPU without network, some scripts (like
while 1: ls -lR > /dev/null) would do the trick.
What does the Spectrum Analysis do in RMAA ? Can we run a long 11Khz wav
file and check the spectrum ? As long as it is plotted with good range
and accuracy, just visually it can be checked for any difference.
@AlexM - Are you noticing the difference between SPDIF and USB output
from the same SBT ? Do you notice difference between Toslink and SPDIF ?


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[ralphpnj]

Reading and following along on this thread makes me think realize
something that I've been saying for quite some time with respect to
digital audio which is that one of the reasons that digital audio was
developed was to lessen or eliminate many of the problems inherent in
analog audio. Based on many of these measurements I would say that
digital audio does indeed lessen or eliminate many of the problems
inherent in analog audio. You know little things like how it is
transmitted, the quality of the cables and ability to reject
interference. Now if only the some of the holdouts in the audiophile
world would just stop focusing on the things that digital audio already
does very well. i.e. lack of jitter and generational losses, and find
the areas where digital audio needs to be improved than perhaps the
sound of digital audio can be made better.


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[Archimago]

ralphpnj wrote:
> Reading and following along on this thread makes me think realize
> something that I've been saying for quite some time with respect to
> digital audio which is that one of the reasons that digital audio was
> developed was to lessen or eliminate many of the problems inherent in
> analog audio. Based on many of these measurements I would say that
> digital audio does indeed lessen or eliminate many of the problems
> inherent in analog audio. You know little things like how it is
> transmitted, the quality of the cables and ability to reject
> interference. Now if only the some of the holdouts in the audiophile
> world would just stop focusing on the things that digital audio already
> does very well. i.e. lack of jitter and generational losses, and find
> the areas where digital audio needs to be improved than perhaps the
> sound of digital audio can be made better.

Excellent point. Maybe in yesteryear (back in the 80's perhaps), there
was much to criticize in terms of jitter, digital cable quality,
ultimate sound quality. But I think since the 2000's at least, "digital
done right" is more of a rule now on the hardware side than an exception
at least with reputable gear (I'll have to try out what a $30 DVD player
from WalMart measures like one day!).

I am actually more concerned about the software side where music is
being compressed to death and no matter what quality gear, it just
sounds LOUD.

Last night I was listening to the recent Emmy Rossum album "Sentimental
Journey" through the Touch. She has a nice voice, the songs are very
accessible with a nice nostalgic feel, the album was recorded with a
live band through vintage mics and supposedly good analogue gear... BUT
the CD has dynamic compression throughout and measures a measly DR7 -
you'd think this was a synthpop dance album! Although it doesn't sound
awful, I can't help but wonder what it could have been if allowed to
breathe dynamically for those of us not listening on iPod's with earbuds
:-(


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[Mnyb]

Archimago wrote:
> Excellent point. Maybe in yesteryear (back in the 80's perhaps), there
> was much to criticize in terms of jitter, digital cable quality,
> ultimate sound quality. But I think since the 2000's at least, "digital
> done right" is more of a rule now on the hardware side than an exception
> at least with reputable gear (I'll have to try out what a $30 DVD player
> from WalMart measures like one day!).
>
> I am actually more concerned about the software side where music is
> being compressed to death and no matter what quality gear, it just
> sounds LOUD.
>
>

1. Call me a cynic but I do think you are more likely to find "digital
done wrong" in the opposite end of the price scale , where lets call it
"unorthodox solutions" are employed .

2. +1 The main problem is the loudness war ,give me good
mastering/recording on cassette tape any day I'll take that over 24/192
or DSD :)


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[Archimago]

Well, the Radio is (sadly) the only Squeezebox device still in
production these days. DAC is the TLV320AIC3104.

I calibrated the headphone output to my E-MU 0404USB.

14434

Not expecting hi-fi performance out of this unit of course and pretty
well got what was expected... Overall, an OK performer in the 16-bit
domain for music streaming duties and incapable of taking advantage of
24-bit audio with only a measly 1dB improvement in dynamic range.

It's a "radio" after all...


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[Archimago]

Well, for posterity and completeness, here is the SB Boom.

Internal DAC is the TI TAS3204.

Like with the Radio, this was connected from the headphone out in the
back --> E-MU 0404USB.

14435

Again, not a hi-fi unit by any means. OK dynamic range but stereo
crosstalk is a bit on the high side. Like the Radio, you can see a
slight improvement with 24-bit data but not much of an improvement to
call it worthwhile.

It sounds great as a stand-alone system which is all I really use it for
in the living room.


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[Archimago]

SoftwireEngineer wrote:
> @Archimago - this is awesome. But it feels like work though (I was
> benchmarking some Big Data code yesterday) :-)
> If you want to just load the CPU without network, some scripts (like
> while 1: ls -lR > /dev/null) would do the trick.
> What does the Spectrum Analysis do in RMAA ? Can we run a long 11Khz wav
> file and check the spectrum ? As long as it is plotted with good range
> and accuracy, just visually it can be checked for any difference.
> @AlexM - Are you noticing the difference between SPDIF and USB output
> from the same SBT ? Do you notice difference between Toslink and SPDIF ?

Thanks Softwire. Quite a bit of work, but as an 'audiophile' + 'geek',
it's been fun :-). Well, it keeps me out of trouble at night after the
kids go to sleep at least!

RMAA consists of measurement of a "signal test track" essentially which
gets played and recorded at the specified sampling rate after
calibration of the playback --> recording system. It's convenient since
all those results I report are collected in one go - usually done within
2-3 minutes. Furthermore, it's quite trivial to run the track through
equipment like the SB devices or even a CD player if I burn the track
out.

RMAA does has an FFT analysis function so yes, you could record a 11kHz
test tone and view it back, but it's not much different from something
like Audacity or Audition in this respect. One package I've looked into
is ARTA for something like this but will have to go thru the manual a
bit more since it's got quite a few tricks up its sleeve including
producing test signals like the Dunn J-Test (11kHz sine + 230Hz square).

Also, thus far I have not purchased the RMAA "Pro" version which I
believe provides more features. Impressed by what the free version can
do already!

I thought about using the 'while loop' also but in the end decided to
see what happens with the more complex multiple TTY instances draining
more memory, net traffic, and each process being of high priority;
presumably adding to the noise from the machine to see if I could
measure it.


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[Archimago]

AlexM wrote:
> Thanks Archimango - good stuff. Nice to know that the analogue outs
> aren't measurably affected by CPU utilisation. That is what my ears are
> telling me too.
>
> As a side issue, I notice that SP/DIF input to my 851c sounds quite a
> bit better than via USB 2.0 High speed (or USB 1.0 Full speed). It
> sounds quite soft and rolled off. I might try to replicate your tests
> (seeing as I have the same gear) and see if this is measurable.
>
> Cheers,
> A.

Interesting observation! Would love to see what you come up with in that
combination.


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