opaqueice;238369 Wrote:
Going into a Benchmark DAC1 I can't hear any change; same with a NOS
DAC I experimented with.
Yeah, but you can't hear the difference between a SB3 and a
Transporter!
--
Patrick Dixon
www.at-tunes.co.uk
Patrick Dixon wrote:
opaqueice;238369 Wrote:
Going into a Benchmark DAC1 I can't hear any change; same with a NOS
DAC I experimented with.
Yeah, but you can't hear the difference between a SB3 and a
Transporter!
Heh, I had exactly the same thought!
R.
auronthas;238455 Wrote:
Thanks for sharing.
So far the sound produced from SF is great, except the low frequecy is
a bit tight. I was told this speaker need a break-in period of 200 hrs
before it delivers its optimum performance.
Hi my experince with sonus faber spekakers are the sound
liffy99 wrote:
Ouch !
But I stand by my own observations (and my other half, who doesn't
listen anywhere near as much as I do, walked in and said that sounds
clearer to a well-known track). I have tried using a linear power
supply as opposed to the wall wart and could not reliably tell a
Patrick Dixon;238470 Wrote:
Yeah, but you can't hear the difference between a SB3 and a Transporter!
Robin Bowes;238489 Wrote:
Heh, I had exactly the same thought!
Can you, blind?
--
opaqueice
opaqueice's
Robin Bowes;238491 Wrote:
I sometimes wish those of you who are so quick to jump down peoples'
throats would take a chill pill and consider that maybe, just maybe
there may be some truth in these reports.
No one doubts the reports have some truth to them - in fact I'm sure
they are
3 threads merged into one.
--
andyg
andyg's Profile: http://forums.slimdevices.com/member.php?userid=3292
View this thread: http://forums.slimdevices.com/showthread.php?t=39714
Hi,
from what has been said I deduce that if you happen to play 24 bit
material and turn the volume down by 48db on the tp digital regulation
you loose exactly the detail that a 24 bit recording has and a 16bit
doesn't. You will end up with the quality of a 16 bit pcm well and off
you go.
Enjoy!
jaysung;238517 Wrote:
Hi,
from what has been said I deduce that if you happen to play 24 bit
material and turn the volume down by 48db on the tp digital regulation
you loose exactly the detail that a 24 bit recording has and a 16bit
doesn't. You will end up with the quality of a 16 bit pcm
ianr;238296 Wrote:
Not sure if I'm just being a niche customer, but does anyone else feel
that Slim/Logitech should consider a Transporter that would handle
multichannel audio (DVD-A etc)?
The problem is, strictly speaking, it is illegal under the DMCA to
extract audio from DVD-A or SACD
jaysung;238517 Wrote:
Hi,
from what has been said I deduce that if you happen to play 24 bit
material and turn the volume down by 48db on the tp digital regulation
you loose exactly the detail that a 24 bit recording has and a 16bit
doesn't. You will end up with the quality of a 16 bit pcm
Mark Lanctot wrote:
The problem is, strictly speaking, it is illegal under the DMCA to
extract audio from DVD-A or SACD onto your PC.
Strictly speaking, it is illegal under the DMCA to write/talk about how
you could extract audio from a protected source.
and there are significant
ar-t;238344 Wrote:
It is inferior, and it is not a myth. You are grossly uninformed.
Galvanic isolation can be achieved by using transformers, althoough
doing so requires some skill on the part of the designer.
Of all the optical methods, TOSLINK is the worst. Single-mode fibre
could be
bwaslo;234946 Wrote:
I have done some interesting tests comparing 24bit/192kHz recorded
music with a 16bit/44.1kHz version recorded from the same mix. I
sample rate converted with the r8brain software so both were at
192kHz, then diffed them. Even with the (I assume) imperfect sample
ar-t;238344 Wrote:
It is inferior, and it is not a myth. You are grossly uninformed.
Galvanic isolation can be achieved by using transformers, althoough
doing so requires some skill on the part of the designer.
Of all the optical methods, TOSLINK is the worst. Single-mode fibre
could be
AndyC_772;238566 Wrote:
What I do find surprising is that anybody designs a DAC that uses the
SPDIF input as a timing reference rather that merely a source of bits.
I've spent some of my spare time this year designing a DAC - based
around the AK4396 as it happens - which makes no attempt
Actually you'd be amazed just how hard it is to hear when a sample is
dropped or duplicated - not that the final design ever actually does
that, of course.
I hope you'll forgive me for not disclosing all the inner workings of
the design right now - it does seem to be a peculiar characteristic of
Andy
I always imagined that by dumping the bits frame by frame into a buffer
and then reading them out aysnchronously but with a very high-rez/low
jitter clock, the end result would be good. Provided that the buffer
never underruns then I see no reason why this wouldn't work. Since the
sampling
Thanks Phil :)
I have a prototype and it works very well indeed.
--
AndyC_772
AndyC_772's Profile: http://forums.slimdevices.com/member.php?userid=10472
View this thread:
CPC;238461 Wrote:
Why do you feel the need to post negative comments about something
you've never purchased or listen to?
Why do you seem to be incapable of understanding my post?
I did not make negative comments concerning the efficacy of wellborne
gear. Therefore it is not germane whether
Andy, that's fine - of course you don't have to discuss it.
Phil Leigh;238596 Wrote:
I always imagined that by dumping the bits frame by frame into a buffer
and then reading them out aysnchronously but with a very high-rez/low
jitter clock, the end result would be good. Provided that the
I must be missing something...the ORIGINAL sampling frequency is a
given...let's say it's 44.1 kHz.
So all you need to do is read those frames out at that frequency. Why
exactly is that so hard? Assuming you never run out of frames to read.
As far as I can understand things, the whole clocking
opaqueice;238603 Wrote:
The problem is that the frequency of the input is -not- given, because
each oscillator has a slightly different average frequency. So your
local clock will never match the one that generated the input exactly,
which means the buffer will eventually overflow or
Mark Lanctot;238537 Wrote:
So to provide a player for illegal formats is not only a bad idea, it
may be illegal in and of itself.
Multichannel audio is better than stereo, period. This has been known
since 1934. My preferred format (B-format) has nothing to do with
legality of ripping
inguz wrote:
Mark Lanctot;238537 Wrote:
So to provide a player for illegal formats is not only a bad idea, it
may be illegal in and of itself.
Multichannel audio is better than stereo, period. This has been known
since 1934.
Your response to Mark's comment is a non-sequitur. He did not
The suggestion that a player for multichannel formats might be illegal,
though, is just silly. There are some formats that would need a
licensed decoder - but they don't encompass the whole world of
desirable functionality. And I'd hate to see Slim hardware forever
tied to two-channel... since
inguz;238623 Wrote:
The suggestion that a player for multichannel formats might be illegal,
though, is just silly.
It's just that the grand majority of multichannel material is on DVD-A
and SACD, and it is not only illegal to rip this material, it is
illegal to even -talk- about ripping this
inguz wrote:
The suggestion that a player for multichannel formats might be illegal,
though, is just silly. There are some formats that would need a
licensed decoder - but they don't encompass the whole world of
desirable functionality. And I'd hate to see Slim hardware forever
tied to
Phil Leigh;238608 Wrote:
I must be missing something...the ORIGINAL sampling frequency is a
given...let's say it's 44.1 kHz.
So all you need to do is read those frames out at that frequency. Why
exactly is that so hard? Assuming you never run out of frames to read.
As far as I can
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