play 96 KHz and hear it on the analog
outputs. When you unplug the loop cable it should go silent.
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the level or using a
higher res ADC isn't going to bring it back.
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dynamic range
(at the low frequencies of interest), while gaining better conversion
accuracy (higher SNR, lower distortion).
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is completely arbitrary and has nothing to
do with amplitude.
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noise floor and it is not realistic to expect that
performance from a $300 device employing a single-chip, single rail DAC
+ output stage.
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in a machester receiver was incorrectly applied to ethernet streaming.
Obviously for Touch there is no s/pdif input so they would have been
playing files over the network, which uses its local (only) clock.
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the same since SNR
will be the limiting factor on a 24 bit DAC.
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and
see if the DAC works.
I have never heard of this problem before so I'm just giving you some
things to try. Where to go next depends on what you find.
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were barely FCC compliant, but the Logitech branded ones were
vastly quieter.
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_phase_ response, and that is not
shown from the simple sweep of amplitude response. It makes sense to me
that this could affect positioning and realism, even if a tonal change
is not detectable. But I'm still not sure why it should be better with
either curve in particular.
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this years ago but I just can't
remember what the deal was exactly. Certainly if they had told me slow
filter is awesome I would have used it.
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certainly are, and in either case the correct type of _cable_ is used,
and that is critical. RCAs connectors aren't _that_ bad anyway - we
manage to run high def component video over them, right?
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your momo;562730 Wrote:
Currently I made very good experience with the BNC out, but I could not
hear clear difference between AES or S/PDIF data format.
There isn't supposed to be.
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is to take advantage
of microphone cables you might have lying around the studio, which are
terrible for this purpose.
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silly that they are even different data formats in the first
place. They just rearranged a few bits in the data structure.
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for the transformers. It is a rough estimate based on the
voltage seen on the secondary side.
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would think
that you're best off letting it do all the resampling in one pass.
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Phil is correct. The grounded plug is for safety, as required for UL
compliance. It ensures that an internal wiring fault could not cause the
metal case to be hot (or cause damage to connected equipment).
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some notes because the
failsafe behavior is excessively touchy and should not be triggered by a
single spurious reading the way it does now.
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michael123;516473 Wrote:
for example, one that does not show me 'overvoltage protection' every
two weeks
see comments in other thread.
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you do have to know what to listen for. Most people probably
focus on the tone and timbre in the foreground which is where mp3 has
the least difficulty.
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the existence of microphonics in general. ;)
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Phil Leigh;514160 Wrote:
I'd call a -144dB null no difference :-)
Yep, me too. Just pointing out where there is still wiggle room for
people to claim they can hear stuff at that level (of course, while
refusing to ABX).
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driven to 0) rather than
floating.
This was different with SB3, where shutting off the DAC was necessary
to reduce the 20KHz noise floor below a level where it would trigger
some amps.
I do not think any difference in component wear should be a practical
consideration.
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with the bath tub business.
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and isochronous USB for example).
In SB/TP, the outgoing data is clocked directly by a fixed crystal. The
incoming data is pulled on demand. There is no feedback from the
buffer fullness to the clock rate.
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it in packets and sudden movements
in an electronic circuit can produce jitter.
It's called TCP. Now, do you have testable claims or do you expect me
to address your hand-waving on the subject?
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by a downstream amplifier,
and that has happened before.
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johnM;510330 Wrote:
I haven't even checked my Transporter to see which kind of connector
they use, hopefully the 75 ohm type.
Yes they are 75#937;
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Can you explain in more detail what you want to accomplish? I don't
think word clock has been tested in this context, but it's not clear to
me that what you're doing is a valid application for it either.
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Valentino;510033 Wrote:
Thanks for replying, seanadams.
The basis is that I would like to syncronize the receiving end of the
effect loop to word clock in instead of the clock embedded in the spdif
signal as I do believe that it will reduce jitter in the system.
It will not. Word clocks
in synchronous loop mode. I know that behringer DEQs do,
but Tact DRCs do not (due to ASRC). It would be a good subject for a
wiki page.
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there is to it.
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Ethernet is transformer isolated and will not make ground loops. Unless
you use shielded cable, which is a dumb thing to do.
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This is next on my list of post-apocalyptic skills to master by 2012.
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not a limiting
factor, I goosed it quite a bit further from the reference design by
upgrading passives, using super regulators and good clocks. No big
secrets there, although other designs will have varying degrees of
headroom for such improvement.
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be a
problem.
Can you use the RCA outs instead?
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Besides the power issue - does that unit even support simultaneous
microphone level on one input and line level on the other?
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gsawdy;503243 Wrote:
Can the wireless card and antennas be removed from the Transporter? Is
there a link to the how-to-do-that info?
TIA, George
It's easy but there's really no reason to. When you're on ethernet it
is totally inactive.
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other reasons is why you should use ethernet if it's available.
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where an external DAC may serve as the master
clock.
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this for one minute, instead of
blindly accepting the dogma of an overpriced audiophile snake oil
vendor, you might begin understand where I'm coming from.
Sometimes I wonder why I still visit this forum.
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!
Not that I know of... and in fairness, Transporter has a word clock
output mode which only does 44.1, but as far as I know nobody has EVER
used it for anything except my own performance testing.
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of everyone which makes it
easy to sell any kind of cure.
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has the advantage that
it isolates the grounds between the two devices.
In terms of signal quality the differences are not going to be huge -
try them both.
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See this thread about analog vs digital inputs on an A/V receiver:
http://forums.slimdevices.com/showthread.php?t=14599highlight=denon
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connections should never be used to actually drive a DAC.
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to understand the most basic principles of DAC clocking. Maybe
DCS does too - or maybe they do get it, but they'd rather sell you an
overpriced, utterly useless but impressively Italian-sounding word clock
box.
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http://forums.slimdevices.com/showthread.php?t=71262
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the best workaround is to use flac to re-encode that one track. It'll
probably work with the default settings.
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-compressing the file using the default FLAC settings, and
if that doesn't work try turning the compression level down a notch.
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the various media.
If anyone actually has a high-end scope and would like to try these
sort of tests I could provide more guidance on how to do it. I think
it's a shame that everyone seems content to speculate about jitter while
so few people actually test it.
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.
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suspect a firmware bug that
can cause a very rare, spurious false positive.
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it is not properly
reinitializing the player when it connects.
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Kellen;485819 Wrote:
All this talk about transmission problems with S/PDIF and USB . can
the Ethernet protocol (TCP/IP) not be used for the transmission of these
digital files in place of the current problematic ones?
I think there are products that do that
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that
is different than what these products do. When you say transport do you
mean a CD player or what?
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the
external processor.
I haven't tested it but I suspect the Behringer DEQ2496 might be
suitable for this mode.
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the right thing and ignoring this
setting, but to be on the safe side it should be off unless you intend
to use it.
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are.
By the way asynchronous (external is master) is always going to work.
Synchronous is an optimization for jitter that only _might_ work.
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AES/EBU is a defective technology, use coax S/PDIF instead.
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the cable handy, used
up all the other inputs, etc) but I don't recommend it.
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suspect neither are the guys who manufacture plastic
toslink adaptors...
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and tested output circuit... you'll
be happy.
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and not the other.
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Pneumonic;483413 Wrote:
Sean, if I may ask. While studying the various DAC chipsets prior to
deciding on the 4396, did you find that most of the serious contenders
outputted a balanced signal?
I have never seen a high end (120+ dB) DAC with single ended output.
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60Hz hum,
ground loops can put other crud in your noise floor that would be
eliminated if you use balanced.
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with a coil of wire in it...
A simple transducer isn't anything per se. It's not until you discuss a
transmission scheme - how it's connected to something, whether one side
is ground and how it is seen by an input, that you can describe the
system as balanced or not.
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/Balanced_circuit
Balanced signaling is not just some mumbo jumbo that audio marketers
cooked up. As I mentioned before it's also the basis for every modern
digital interface, and not just for interconnecting cables but (for high
speed signals) even at the board and die level.
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uptime has been reset (i.e. it crashed and rebooted).
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offset by that much.
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bad - you're missing the
point of balanced signaling. It's TWO complementary paths, but neither
one is more complicated than its single-ended alternative, and having
two complementary paths is what gives you the common mode noise
rejection.
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That is too bad. I wish I could be of further help.
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supposed to come into play when TP is
being used as a DAC.
Unfortunately, I think you will probably need to send it in for a main
board replacement.
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), then it's bad. You might even convince Slim to send you just
the module and save everyone on shipping.
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maybe it's using a surround encoding? Any option in system preferences
- sound to change it to PCM?
Other than that, maybe try playing it into an AV receiver and
confirming that it works and comes up as PCM.
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.
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Sorry but I can not imagine any reason for this not to work!
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What makes you think it's the PSU? If the CPU doesn't boot for _any_
reason, you will get the same behavior. I would recommend sending it in.
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passes everything through ASRC
processing even if the sample rates are nominally matched. I remember
testing this when developing the TP effects loop feature - you always
have to let the Tact become the master clock.
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alone.
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unit.
Actually, given the way the bathtub curve works, a well-burned-in
Transporter that still passes factory testing is statistically _less_
likely to fail.
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feeding. If it's going a short distance straight
into an op-amp buffer then you're probably fine. But if it's got some
capacitance or noise nearby, then you may have a problem.
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dB of digital attenuation in the TP as needed.
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prefer.
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to the chip in
Transporter - although it lacks the Jung power supplies and low-jitter
clocking which will give TP a distinct advantage. If you want to do a
serious test it would be necessary to match levels using a voltmeter on
the pre-outs.
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to this sort of
thing.
A few things you could try:
Are you using the wii's 802.11? Use ethernet if possible.
Connect the wii with optical s/pdif if available.
Clamp ferrite cores around the wii's cables. Try them near either end
of the cable.
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amcluesent;439601 Wrote:
But I feel that this isn't so good due to impedance mismatches.
Nonsense. Line-out RCA connections are not even impedance matched in
the first place. It's perfectly OK to use a splitter for this.
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that the impedance of the
trigger input is high enough that it doesn't get loaded down too much.
If that doesn't work, use a 50mA PTC (resettable fuse) instead.
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what you're getting at here. s/pdif uses only a continuous
clock signal that is embedded (manchester encoded) in the data. It does
not rely on the kind starving or not flow control you're imagining.
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his statement is quite disingenuous in extrapolating that
observation to current models, or the SB clocks in general.
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timequest;427581 Wrote:
I really want to see if I can mod the Duet receiver with an I2S output.
Is that doable?
Not much you need to mod, the signals are there.
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too hard. They have a wattage rating but again this is for
some test signal, probably pink noise, and is only an approximate guide.
For real music you just don't know its power level (at the relevant
frequencies most likely to damage the speakers) in advance.
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you should instead be
using -20dB analog attenuation.
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This is not feasible, as Transporter is doing a simple electrical
pass-through from the s/pdif receiver to the DAC. From in-to-out there
is a fixed latency of less than a millisecond.
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Never mind, the inversion option is only on Transporter.
Another thing to try is to test if it's a timing or performance issue.
Try turning the spectrum visualizer on and off, and see if that affects
the behavior.
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Try it with the analog outputs instead. Need to determine if it's an
s/pdif issue or something with the decoding.
Also try playing the files on a PC - make sure it's not the source
material.
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follows, k=7 = 001 follows.
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