ralphpnj wrote:
Nor did Mr. Swenson give any numbers or measurements regarding the types
of errors these green markers are supposed to correct. His response was
all very nicely written and with a seemingly clear logical and
scientific foundation but all beyond the prose and everything
Archimago wrote:
Greening the edge of CD's was all the rage around the time I first got
some $$$ to buy audio stuff.
Start date: ~1990
Reason for the buzz: Some vague theory about stray red laser light
bouncing around the polycarbonate. An article in ICE (International CD
Exchange)
Archimago wrote:
Fantastic info. Thanks John.
BTW: Are you running room correction software on these? Curious what
plugin...
I am not running room correction software, but I am running upsampling
in Squeezelite using a filter spec that a friend and I developed for the
DAC chip we are using
Archimago wrote:
John, which Wandboard are you guys basing the CSP design off of? Duo,
quad?
Thanks...
The software is written for both the dual and the quad. Everything runs
fine on the dual, I've had a dual running LMS, playing 192k files,
filtering in software, outputting to USB and it
I have written a series of articles on AudioStream that go into
technical details on how differences can get through reclocking.
For use with a USB input you might want to try the Community Squeeze
project. ( www.CommunitySqueeze.com ) runs on a Wandboard (small Arm
computer) which does VERY
marcoc1712 wrote:
John,
could I ask You witch USB DAC are you using? If is kind of a KIt or a
commercial product, could you point me to the source?
is a kind of the one you'll be using in CSP1?
thx a lot.
p.s.
I could not understand why if you upsample at i.e. 352.8 you still
Julf wrote:
So how do you measure what a SET does well?
I was wondering if someone would ask that!
The part that seems to make the difference is a monotonically decreasing
harmonic structure. The third harmonic is a little less than the second,
the forth is a little less than the third etc,
I'd like to reply to some comments in this thread, particularly to the
ones about SETs and NOS DACs. These are not in the same category as
electret cream, they are real life engineering tradeoffs. I'll give some
examples of both.
I'll start with the SET. I have built MANY amplifiers over the
The Fez wrote:
Hi Guys -- Had a SBT for years now. Really nice product. Just wondering
if anyone else has a Audio-gd DAC connected via USB. I have the EDO
installed, but the SBT doesn't recognise the DAC without a USB hub? I
was of the impression the AGD DAC'S can connect direct to the
I started this because I was using the upsampling built into
Squeezelite(libsoxr), not using SoX in LMS so I can't offer any clues as
to how to get that to work properly. The Squeezelite resampling option
is not the same as in SoX the program arguments, although the underlying
code is the same.
edwardian wrote:
Cool. Thanks for the info. Do those percentages include FLAC decoding,
or are you feeding it PCM?
I'm usually sending flac over the network these days since I don't have
a new enough server to handle 176 and 192 pcm. The above numbers were
for sending flac at 44.1 over
Hi Pippin,
to your underlying question: why do the filtering externally rather than
in the DAC chip? The answer is I don't know. When I bypass the
internal filter and do it externally using a basic simple filter it
sounds much better. And it's not just me. I've done this in blind
fashion
pippin wrote:
So what you are saying is that if you use these kind of DACs with real
(that is: unfiltered) HD material like actual HD recordings you will
have all the aliasing down there in the audible band?
This means you have to actually run real HD material through a filter
to get a
pippin wrote:
But then you've got all these high frequencies still in there on the
analog side. Isn't the biggest problem with the high frequencies that
power amps (especially ones filtering through intermediate frequencies
like Class D amps) tend to create artifacts if you have signal in
Wombat wrote:
Many 44.1 material once was at a higher samplerate and there was already
once a choice what filter to apply. How can you know that a chosen
filter doesn´t only do better because it fits more to the filter that
was applied before? I did read to much nonsense about the sound of
As promised here is post number two on external upsampling. This post is
going to focus on the upsampling capability recently included in
Squeezelite which is based on the SoX resampling code. These filters
can also be used offline with SoX, but the command line arguments for
SoX are very
In recent versions of CSOS Triode and JackOfAll have added upsampling
capabilities to Squeezelite (at my request). I wanted to start a thread
here for discussing sonic differences people hear between no upsampling
and different upsampling parameters.
I'm also going to give some history of my
There actually IS a technical reason why DSD may sound better than PCM,
even though it is technically inferior to hi-res PCM: lack of digital
filters. I have done extensive study on this and have come to the
conclusion that ALL DAC chips which have internal digital filters have
not correctly
jh901 wrote:
The Transporter's clock is vastly inferior to those found in today's
best DACs. Vastly.
Anyhow, is asynchronous USB superior to S/PDIF? In any way? I've
pointed out that slaving the Transporter, for example, to a master is
not possible. If the Transporter had asynchronous
SB3 supports 44.1 and 48. Natively the Touch supports 44.1, 48, 88.2 and
96. There is an applet for the Touch called EDO (Enhanced Digital
Output) which will pass 176.4 and 192 on the digital outs, AND allows
you to connect a USB DAC to the Touch and will go up to 192 if the DAC
supports that
There is only one situation where the word clock in is useful: you have
a DAC with a very low jitter internal clock AND it can output a word
clock derrived from that internal clock. You connect one of the digital
outs from the TP to the DAC and the word clock from the DAC to the TP.
This slaves
First on the term apodizing: different people have radically different
meanings for the term, so it's very difficult to figure out what any
body means.
If I remember correctly the original use for the term in regards to
audio digital filtering was about a non-brickwall filter applied to
higher
When you feed an external wordclock into a Transporter it has to use a
PLL to generate the high frequency master clock used inside the TP.
This PLL is going to have much greater jitter than the crystal
oscillators already in the TP.
The ONLY reason to use a wordclock with a TP is if you already
Note that I specifically said that the tests I had run were done in the
same room as the stereo system, usually 6-8 feet away, NOT several rooms
away.
These tests happened about 5 years ago when I took my SB3 over to a
friends house to show it off, how nice it was to use and how good the
sound
chill wrote:
John
I can't tell if you're just being mischievous, but if not, are you
seriously suggesting that the mere presence of a powered-up computer in
the same house as the hifi will have an audible effect? That's a bit of
a blow for the whole computer-based audio industry. But
Turnandcough wrote:
Sorry if this has been covered but I couldn't find anything.
24/96 is more than enough for my ears so I'm not really interested in
the 24/192 aspect here.
1) Can I use the Touch USB output with an Audiophilleo? Do I need to
install EDO?
2) Theoretically(or
Jeff Flowerday wrote:
I'd be interested in John's opinion on these transformer based
converters.
These transformers can be quite decent. This is how I would do an
AES/EBU output. On anything I do with an S/PDIF output I provide a 75ohm
BNC, if the user wants to go to an AES/EBU input they can
The Transporter will not do 176 or 192 on it's outputs. If you play such
a file the server will down sample 176.4 to 88.2 and 192 to 96.
The Touch will output 176.4 and 192 on it's digital outs if you use the
EDO applet from Triode. EDO also allows you to hook up a USB DAC, which
can also
chill wrote:
You make a good point. I stand corrected.
So do you think the OP is interested in the THE simple test that will
demonstrate that SuperQ's suggested mechanism is the correct one? You
know, the 'just pull the plug' test.
But that test is not necessarily defintive, it only
konut wrote:
The prevailing wisdom asserts that no RCA termination will give a true
75 ohm result. Then again if the source and DAC were designed to be
impedance optimized with the RCA jacks then this wouldn't really matter.
Unfortunately impedance was not a design characteristic of the RCA
I knew someone was going to ask this!
The above post was actually an oversimplification, in actuality the
impedance can vary with frequency. Its similar to saying a speaker is 8
ohms, very rarely is it 8 ohms at all frequencies in the audio range.
Some reviews will publish the full graph of
jh901 wrote:
2) I'm using the coax output of the Transporter for now, but I'm
switching to AES since the rest of my system is balanced. Are any of
the Transporter outs considered superior (for sound quality)? Most
importantly, is the digital output from the Transporter being clocked
cfraser wrote:
Hey Ron, is there any more handy info about this style/brand of server
that you'd recommend? It sounds like exactly what I want, especially the
low power consumption for 24/7 op. And a small useful project just
when I'm out of them. Thanks.
I also have a FitPC2 as one of my
lake_eleven wrote:
I deleted the existing jive_alsa. Copied 0.6 version, changed the file
access, then rebooted. Am I missing the sync, Not sure what is this
sync step.
When changing files the changes go into a RAM buffer rather than
immediately onto the disk (flash in this case). When the
Triode wrote:
It has been reported elsewhere that tuning the kernel module parameter
nrpacks may have an impact on sound quality with EDO when using a usb
dac.
This can be achieved by:
echo NUMBER /sys/module/snd_usb_audio/parameters/nrpacks
Please note that if you try this tuning
JJZolx wrote:
How does jitter get through an asynchronous interface? Isn't that a bit
like talking about jitter on an Ethernet connection?
Good question, unfortunately things are not as simple as it seems on the
surface. The supposition goes that as long as you have a local clock,
the only
The USB isolators block a potential path for a ground loop through the
ground of the USB connector of the computer (Touch in this case).
Whether this is necessary will of course depend on the computer and DAC
and how the power supplies are hooked up. There has been so little use
of USB DACs with
lake_eleven wrote:
John,
I have placed my order for HRT Music Streamer II+, hence very interested
to learn the changes to your system with HRT. I have:
- Installed EDO
- Installed TT3.0 over it
- Changed priorities in TT file to Logitech default
- Made the changes suggested by SBGK on
totoro wrote:
Actually, it's not all that clear to me that what you're saying makes
any sense, either. Even if you have a bunch of threads running,
depending on the types of locks they're using underneath (spinlocks or
some kind of sleep locks), if threads are waiting on locks, their mere
The way EDO works when selecting the 192 digital out will use less
memory and will result in less processing going on. (please note that
this output will do all sample rates up to and including 192, not JUST
192). What Triode has done here COULD have been done with the original
S/PDIF driver (and
I have a HRT Music StreamerII (not the +), it works very well with
Triodes plugin and a hub. Adding TT3.0 significantly improves the sound,
with this combination it's getting astonishingly good. (it's not the
best, my homebuilt DAC still blows it away). The II with TT3.0 is
sounding better than
Hi Rolf,
have you tried listening to the analog outs of the two Touchs? My
experience with putting lower jitter clocks in SB boxes is that it does
affect the analog outs. I haven't measured the clock mod that was done,
but I HOPE that anyone who is making a 3rd party clock board is going to
be
cliveb wrote:
John, thanks for trying to explain, but I'm still confused. Your very
first paragraph is completely at odds with what I have always understood
to be the purpose of oversampling in a DAC:
I thought that the effect of oversampling (ie. zero-stuffing) was that
it moves the
filter at that sample rate.
John S.
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in that ultrasonic carrier.
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NoRoDa;699015 Wrote:
Interesting findings yesterday!
HWmods made no difference, but at least did not sound worse.
Regards
Which hardware mods had been implemented? Were you using coax or
optical to the DAC?
Thanks,
John S.
--
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by
the internal digital filters. Of course then you have aliasing all over
the place. For some people its worth the tradeoff.
Of course the proper solution is to use proper digital filters so you
can have the best of both.
John S.
--
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it just right and getting blown
away.
BTW what do I like? I prefere #2 in the list above, its hard to do
right but the results are astonishing if you do get it right. Of course
then you have to deal with the ground plane issue, but thats a whole
different ball game.
John S.
--
JohnSwenson
a kernel which will play 192 over the
S/PDIF output of the Touch.
BTW the Touch has always been able to play 16 bit adaptive USB, its
been the 24 bit async that's been the problem.
John S.
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,
dropouts, cliks, pops, etc) I would have no qualms about streaming
flac.
John S.
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soon is pretty low.
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coupling
cap in the chain.
I was specifically looking at an article that was analyzing what
happens with an unbiased polarized electrolytic while varying the input
voltage from .1V RMS to 2V RMS. I don't remember exactly where it is, I
can't find it right now.
John S.
--
JohnSwenson
ways you can do it.
Its still not as good as getting rid of the capacitors all together,
but its much lower distortion than what you get out of the box, and you
don't have to do anything to the Touch itself.
John S.
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That article is for a properly biased electrolytic, an unbiased one is
MUCH worse, which is the situation we find in the Touch.
John S.
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around will either make the connection normal, or no sound
at all.
It could also be a broken solder joint in the connector (cable, or
Touch), again usually wigling things around will change it.
John S.
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S.
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want to go that far!
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filter. The only ones I have found that
seem to do a pretty good job of their internal filters are the Sabre
chips.
Well there you have it, some of my exploration of NOS DACs.
John S.
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JohnSwenson
JohnSwenson's Profile
formats by using a browser to login to the LMS web
page address:9000 then click on settings, then advanced. Then click
on the pull down arrow on the left side and select file types.
John S.
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JohnSwenson's
changes, thread priority
changes, ALSA configuration changes and turning off things.
John S.
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should be able
to take some of the modifications that do not require a reboot and put
those directly into Triode's framework without having to use the tt
interface.
John S.
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together, some of which require a
reboot and some which do not. It should be possible to run the ones
which do not require a reeboot under Triode's framework.
John S.
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. There are a lot of factors to take into account.
John S.
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the scripts do
that.
Any takers? Any discussion? Does this satisfy the DBT rules?
John S.
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part of quite a few tests that do just that, I was trying to
point out that this type of test fosters the type of testing I find
less valuable and there is a different way to do it.
John S.
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JohnSwenson's
.
John S.
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, when color sets started coming out people would frequently
get a portable color model that had to sit on top of something, what
better place than the old BW console. No need to buy a TV stand.
John S.
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JohnSwenson
they
can wind up increasing certain incoming noise frequencies.
John S.
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that seem to belive that that sticking a
pulse transformer on a S/PDIF stream automatically cleans up all
noise.
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means more code gets swapped in and out of main
memory.
This is something that is really hard to measure since OSes don't
usually give you a processor cach hit rate graph you can look at. (of
course the process of monitoring it would decrease the hit rate!)
John S.
--
JohnSwenson
with.
Note that at no point did this DAC sound BAD or obviously distorted,
it just made music sound significantly more real as the changes were
made.
John S.
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.
So I would say that improving the clock is actually quite a worthwhile
mode if you can do it. Unfortunately there is VERY little room in there
to do it.
John S.
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. Unfortunately there is no tool you
can run that gives you a graph of processor cache misses, it would take
a logic analyzer looking at the memory bus. I think that focusing on
processor utilization is a red herring and not worth worrying about as
long as it doesn't get too high.
John S.
--
JohnSwenson
an hour and been on for 10 hours.
John S.
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this is with a good
external supply and output caps removed) But with S/PDIF things seems to
be a little more challenging to get really really good results.
John S.
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to the regulator, just a pure 120Hz
sine wave which just about anything can deal with.
There are other advantages but these are the two main ones.
John S.
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to accidentally turn on one of these modes without
realizing it.
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I have posted on these forums uses a CLC filter which loads the
transformer with an almost pure sine current load. It also feeds pure
120Hz to the regulator which makes it's job MUCH easier, it doesn't
have to deal with all those high frequency harmonics.
John S.
--
JohnSwenson
enhancement.
On the USB dac fron in general I still have not got it completely
figured out, I'm still getting ocassional xruns no matter what I do.
I'm starting to belive its down inside the low level USB driver, but
I'm not sure about that.
John S.
--
JohnSwenson
for this amp,
you cannot tie the negative outputs from the two channels together. As
long as you have two wires for each channel and don't connect then to
each other or to anything else there wont be a problem.
John S.
--
JohnSwenson
. Once you can get in you can modify the startup
scripts so you don't need the USB/SD anymore.
You can do a lot of things with this script if you want.
John S.
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opinion as
gospel, there is such a huge variation in equipment so that what
sounds the best to one person is almost guaranteed NOT to be the best
for you. You HAVE to do the testing for your self with your system.
John S.
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.
John S.
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with optical, although because the common
implementation is so poor to begin with noise on the signal is rarely
noticed.
John S.
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connector. Quite a while back there was an
attempt to do this with the ST optical interface, but it never really
caught on. Until someone starts making a GOOD optical receiver with the
same TOSLINK connector not much is going to change.
John S.
--
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is the union of those two
filesystems. A factory reset deletes everfything in the modified
filesystem, thus you see the as delivered firmware.
A firmware update deletes some of whats in the modified filesystem,
thus some changes will go away and some will stay.
John S.
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.
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which
determines on the fly what the process numbers are, you just need to
add detection of the correct jive_alsa thread to this script.
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be an interesting
test to try running jive_alsa at a higher priority (say 55) and then try
running s/pdif in the 50 to 60 range.
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View
they get played on the main
channel they cause all kinds of crackling etc because they are at a
different sample rate than the music.
As has been mentioned the solution is just to turn off all effect
sounds.
John S.
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. This sort of
gelled for me, like when you are adjusting VTA and get it right.
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to get this figured out.
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, they work well, are cheap,
and readily available)
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sounding significantly better.
But I don't think that change is going to happen with the squeezebox
system.
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.
On the Touch you need to SSH into the Touch (use PuTTY on windows) and
cd /var/log
less messages
When done type q then exit to get out.
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res 24
bit playback to work and its almost there with the HRT. It still once
in a while, but it's much better than it was.
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that.
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a particular DAC.
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