Wombat wrote:
> I knew ULTRA makes guys horny, look further, it is out there!
Yes it does ;-) Can't seem to find it though, a link would be
apreciated. :p
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Allow aliasing above the pass-band",
" -b 74-99.7 Any band-width %",
" -p 0-100 Any phase response (0 = minimum, 25 = intermediate,",
" 50 = linear, 100 = maximum)",
};
Nothing about an ultra quality mode?
-
and technical solution
for this, thats all.
Ronald
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d file.
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Apesbrain wrote:
> Yeah, I'm just going to go ahead and call bullshit on this.
Having an opinion is fine, but getting personal does not help a lot most
of the times.
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ed tracks were indeed copied to /tmp.
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tracks now play from RAM memory, resulting in a more stable stereo
image. And yes that is very subjective ;-)
Maybe someone else likes to try this as well :-)
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Personally I still prefer the polyphase upsampling from sox 14.2
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Can't wait to try. So thats exactly what I will do, going to try right
now ;-)
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Bummer... There only seems to be a windows version available :-X
when will there be a new beta of the linux (debian) version?
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andyg;568593 Wrote:
QA is going to start testing TP firmware 84 today for 7.6 beta so it
should have this in there too. :)
I had totally forgotten about this topic. Thanks a lot Andy for doing
this! :-)
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Simply remove the wireless mini pci card from the transporter, thats
what I did ;-)
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want to change anything on the Transporter anymore.
Ok, so I wasn't the only one who was wondering. I think slow roll-off
might be sound better ( but measure worse).
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Hi,
Would it be possible to make the choice between slow and sharp rolloff
filter in the AK4396 (transporter) user selectable? Personally I would
be very interested to compare (subjectively) the difference.
Ronald.
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Yes, balanced sounds better, even when using one leg to an unbalanced
input of an amplifier. This is explainable because one opamp/filtering
stage is omitted this way compared to the unbalanced output of the
transporter.
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Today I also tried some 192...@24 bits flac files on my transporter
(squeezeboxserver 7.4.1), they play fine.
Are you sure these are being downsampled?? The AK4396 is pefectly
capable of handling 192 khz samplerates.
Ronald.
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purposefully by the
device(equalizer or re-sampling).
Don't forget that these 0's and 1's have to be clocked by a very stable
clock, that's where the difference is in sound between digital outputs,
not the 0's and 1's.
Ronald
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It was clear (for us) the rate method was sounding worst. Rabbit sounds
quite good and with some advantages over polyphase (cleaner highs, very
detailed and precise) but polyphase sounded the most musical with a bit
more body to the sound.
Ronald
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michael123;516615 Wrote:
I think polyphase returned me some 'punch' I was missing with the 'rate'
effect
Yep, it has more body (in a positive way) to the sound and almost
sounds a bit tubey :-)
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PM sent :-)
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Yesterday me and an audiophile friend compared the rate upsampling with
minimum phase filtering against polyphase and rabbit upsampling. We
clearly preferred polyphase AND rabbit over rate with minimum phase. Its
a shame those 2 upsampling methods have been deleted since sox 14.3
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of the AK4396 states that it does 128x oversampling,
which is part of the sigma delta design in my opinion. There is nothing
in it about upsampling But ok, I do of course agree that whatever
parameters used with sox, there is no way to work around the existing
filters in the DAC.
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This topic mentions creating a wav file representing an impulse. Does
anyone know how to create such a sound file?
I would like to experiment a bit as well with comparing impulse
responses ;-)
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michael123;512671 Wrote:
nice
plz let me (us) know what did you get from each filter, and what did
you like the most
Sure I will! However, the differences were small (or maybe I was tired
yesterday..) and I was not able to draw any good conclusions.
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DaveWr;512912 Wrote:
Still don't forget the up-sampling filters in the Transporter
Dave
Which upsampling filters? As far as I know the transporter does not
upsample at all (?)
If you ment the DAC filtering, then yes they will do their job as well.
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I am also listening (as we speak) to the difference between minumum
phase vs lineair phase, limited bandwidth (85%) versus less limited
(95%), polyphase versus rate -v. Pf, drives me crazy ;-)
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Personally I would love to have the transporter in an old fashioned hifi
component look (but with the display of course).
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A better analog output stage. After rebuilding mine with discrete opamps
(burson audio) it sounds significantly better (more 3D and more analog,
more resolution and less agressive highs).
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Thanks for all the info. I am actually more interested in the digital
output of the SB3+ since I already got a very good DAC (PS Audio
Ultralink II).
I'll certainly give the SB3+ a serious thought !
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Nuuk;320638 Wrote:
Try here http://www.at-view.co.uk/at-tunes.htm.
Thanks a lot!
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. Therefore the
effect of the mic cal file is the opposite of what you would expect.
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inguz;230275 Wrote:
Well, there's no way to get a really true reading with an uncalibrated
microphone. And I don't have a calibrated mic. So that's my next
step...
I got mine calibrated a while ago and my advice would be: It should be
the first step! :P
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I just received my DIP Classic a few days ago, its between SB3 and
Apogee Mini Dac now and makes a clear improvent. The digital edges
around voices are gone now :-)
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to insert some silence before the impulse response created
from ImpulePrep.exe, the octave scripts expects to have at least 512
samples (if I recall correctly) before the impulse center.
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.
However I am a bit puzzled by the fact that 2x 22.5792 = 45.1584 and
not 44.1 :P
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. The
16MHz is
for USB.
- Toby (hardware/software guy)
Hi Toby,
The ratio might be correct, but the absolute frequencies are weird
aren't they?
Ronald.
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Toby Dickenson;222871 Wrote:
tingtong5 wrote:
245KD6H (labeled X2 24.576MHz)
The ratio might be correct, but the absolute frequencies are weird
aren't they?
24.576MHz = 512 * 48kHz
How stupid I am : Mhz and kHz , not the same thing :-X Say no
more :(
;-)
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sample rate converters are
exactly of this kind and are one of the most common cause of poor
quality impulse response measurements and consequently of correction
artifacts.
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if there are resampling artifacts in the
sweep recordings?
(Tingtong -- I was reassured by your mic calibration plot -- roughly
similar to mine.)
Simply open it and see how many oscillators are in there and what
frequency is printed on them!
Yes they look quite similar :-)
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(impulse response)..
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muski;221338 Wrote:
Tingtong --
Where did you get your mic calibrated?
muski
In germany (I live in holland):
http://lasip.hifi-selbstbau.de/cgi-bin/shop/shop.cgi?shop=product=Kalibrierungcart_id=2692111.29825
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to measure at 44.1 kHz).
This is not good (the resampling) so I am starting to look at buying
another soundcard now :P
Any hints for an USB or pcmcia card with a native 44.1 kHz oscillator
would be welcome!
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will have to read the manual ;-)
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-corrected low-frequency response..
Life ain't easy ;-)
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-14kHz
range...
Nothing much else is happening up there. Guitars are gone by 6KHz (at
best).
Sorry for the confusion. I ment that I am using a rollof (beginning at
400 Hz) that eventually reaches -8 dB at 20 kHz..
I fully agree with what you are saying !
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Unfortunatly I can't because I do not have the means to measure jitter.
The improvement I notice is therefore purely subjective :P What I can
do is to compare it in an AB test to an unmodified SB3 but then I have
to invite someone over sometime that has one :P
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harmonic;220333 Wrote:
Tingtong duck for cover or run
Why is that? :P
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harmonic;220351 Wrote:
You are not doing enything wrong at all, but none measurble mods= flame
war
At least I am not claiming anything, except for the fact that I think
that this is subjectively an improvement :P
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an improvement after that mod it must be from lowered jitter or it
might be a placebo effect. I think it is the first, but I just can't
prove it ;-)
Ronald
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jitter clock, in that
case it turned out that the unmodded SB3 sounder better :P I think this
mod hs a higher chance of sounding better :P
We'll see..
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! But we did not measure.. However for me the effect was
significant and beyond any doubt a big improvement.
By the way I think the SB3 is great, my remarks in my previous post
were with high-end results in mind, which of course cannot be
expected of a device in this price category.
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!
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That's why I choose to use a flat curve correction file for now instead
of the drc correction curve. Guess its time to have my mic calibrated
:)
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measured
one ecm8000 mic which does not mean is it valid for all of them..
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it as well (got a copy myself).
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After buying a decent sound card I get good measurements now and am
enjoying a proper working room correction now, which really is a huge
improvement over non-corrected sound :-
I am a happy guy :)
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Marley3;206373 Wrote:
There must be something I'm missing...
Jitter.. As said before, it depends highly on your dac wheter you will
here these differencs or not.
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microphone together with the behringer tube mic amplifier and a cheap
usb soundcard. I guess I might have to replace the soundcard to start
with.
Regards,
Ronald
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inguz;205316 Wrote:
tingtong5: yes, this looks absolutely normal. The best startingpoint
for a target file is usually the bk-5-spline rather than flat, but
anyway try listening with flatness values around 3 to 5, see if this
brings back some of the frequency balance you're used to.
Thanks
.
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/Tools/ImpulsePrep/
This evaluation version will expire 11/1/2007 12:00:00 AM
Processing left measurement (left.wav)...
Thats all, no output anymore afther that :P
When I execute ImpulsePrep on windows it does do something but then I
have no DRC :P
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?
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Inguz. I used the ecm8000 correction file
and the flat.txt as target response.
What do you guys use as mic pre-amp ?
Best regards,
Ronald
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.. Hopefully
I can get my hands on one to make the comparison :P
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of the
coax cable's shield.
I suppose this is a big sin since we are talking high frequencies here
..
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I think was a mistake..
Yes I do have a scope, though it is a 10 MHz one..
The clock looks more like a sine wave then a square wave but I assumed
this is because of the scope's limits in being able to display
frequencies 10 MHz..
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So I guess I need to try again and do it the right way this time :P
Now I have to find a good ground reference point on the circuit board
somewhere near the pad which serves as the external clock input..
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of
the SB3, but not in the proper way..
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Patrick Dixon;190250 Wrote:
I can't quite see from you photos, but you do have a ground reference
between the SB3 and the tent clock board, don't you?
Yes I do... :P
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this the asynchronous reclocking (of data
and wordclock) would then automatically become synchronous reclocking.
I would have to attenuate the clock signal that goes to the SB3, since
the tentlabs XO has a 5V output. I will use resistors for this.
Regards,
Ronald
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