Here is what I use: http://www.outlawaudio.com/products/2200.html
Balanced in, clean, low noise floor, powerful enough to drive just
about any speaker.
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On-wall speakers? http://www.magnepan.com/_mmgw.php
Just be prepared to use a high current amplifier to drive them. My
current rig uses a DAC1 PRE into Outlaw Audio 2200 mono blocks driving
MMGs.
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Timothy Stockman
for you.
http://www.magnepan.com/_mmg.php
http://www.outlawaudio.com/products/2200.html
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the breadboard
(toslink only) and FIRST PCB stage. Normally, the general public
doesn't see my designs until about the third revision of the PCB. I'm
sorry I let you and others see an unfinished design. I'll immediately
remove the link.
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Timothy Stockman
Semiconductor datasheet. My coax output
circuit is copied from there. I have not tested coax, only optical,
on my prototypes, so maybe they made a mistake. Or maybe they know
something we don't.
http://www.cirrus.com/en/pubs/proDatasheet/CS8405A_F2.pdf
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Timothy Stockman
made the PCB layout, I never built one with the coax parts
populated (I always used Toslink in my prototype boards), although the
transformaer output circuit is directly from the Crustal Semiconductor
datasheet, so I wouldn't expect it to fail horibly.
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pfarrell;350525 Wrote:
Timothy Stockman wrote:
I had the most years of experience with a Collins 21A 5KW transmitter
at WBAA radio. The modulation transformer (a Thordarson True
Fidelity model) and modulation reactor were so big and heavy that
they
sat, along with the 3 power
the wall adds $1000 to the sound. :) Now all
I need is a Transporter to go with them. (I think that will have to be
my next major purchase!)
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(384 x FS) PLL would
even stay locked!
http://proxy1.fielddiagnostics.com/~tps/i2s/
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a reconstruction filter with a much gentler slope, which would
typically have superior sonic characteristics. Modern DACs achieve
most of the benefits of higher sampling rates by using oversampling
digital reconstruction filters.
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Timothy Stockman
on the hard
drive, then used DVD Audio Extractor to read the hard drive copy and
make 24/96 WAV files.
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to the
original levels and people didn't like that after hearing them louder.
They didn't realize or care that they were listening to music with the
mids way low and not accurate.
This is why Dolby was often jokingly referred to as Dullby. :)
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is an
effective band-aid. I also carried a couple medical power isolation
transformers for situations where it was more desirable to put the
transformer in the mains circuits rather than the audio path, but the
effect was exactly the same.
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Timothy Stockman
level events such as reverb tails, etc. But still,
with 24 bits of precision, there's still a lot of fudge factor to keep
the quantization distortion which occurs as a result of the simple
volume adjustment pretty much inaudible at reasonable attenuation.
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Timothy Stockman
stray AC currents in the first place.
For instance, a power transformer with non-concentric windings or
Faraday shields.
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equipment, the more likely cause
leakage from the AC power injected into the cable to power the line
amplifiers. The subscriber taps remove most of the 60 Hz AC, but as
was pointed out by the previous posters, an RF isolation transformer
will fix the problem.
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Timothy Stockman
. Note that, being an EQ
operation, de-emphasis is a lossy operation. Maybe it would be good to
use something process the results with something like lossyWAV before
encoding to FLAC, which will make a better, dynamic choice as to how
much calculation residue to retain.
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Discovered Again - Dave Grusin (Sheffield Lab CD-5)
Gaucho or Aja - Steely Dan (DIDX-56, DIDX-57, IIRC)
Are good ones... And there are a few recordings I made myself at
various live music events, either as demos for the bands or direct off
the board when I was mixing the PA sound.
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Timothy
as I was running it
for my own use as a tool to evaluate the mix.
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of the medium, which possess the magic, there are many
average recordings, and there are far too many poor recordings...
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understanding?
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to using the 16-bits resolution the CD provides. IMHO
what is important is that the digital volume calculations are done in
greater than 16-bit, 24-bit should suffice, and that the residue of the
calculations not truncated to 16-bits.
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/BNC_50_75.html
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the metal
strings, would get a shock on his lips if they touched the metal
windscreen cage of the vocal mic. A medical isolation transformer
always fixed that problem, as well as the hum it usually caused.
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examples of this one is likely to
encounter is digital dither between the tracks of a CD. If you zoom in
on it in an audio editor, you'll see that it often varies between 0 and
-1. This signal is definitely NOT centered on zero, thus it is
asymmetrical with regard to 0.
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-bit recording can indeed have
16-bits of resolution, not 15-bits (1 bit reserved for the sign). Yes,
the sign bit is there, but it is part of the 16-bit resolution, too,
mainly because of minute waveform asymmetry.
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recordings,
generally the bottom few bits are filled with noise.
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changes my opinion.
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it makes the
most audible (to him) difference, if, indeed, that is his goal.
Hmm... Maybe I can find an audio dealer who will imagine that I paid
him $1000 for a speaker cable so I can imagine I hear a big improvement
when I use it in my system... :)
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solution to this is ReplayGain, which will recognize that hdcd.exe
has reduced the level by 6 dB and make a corresponding adjustment to
the ReplayGain value.
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for playing a mix of discs decoded by hdcd.exe and non-HDCD
discs in shuffle mode. But, given the wide variance in mastering
levels over the years, ReplayGain is often fairly necessary in shuffle
mode, anyway.
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Timothy Stockman
Microsoft
secrecy surrounding the official specs makes me wonder if any of us can
truthfully say that it results in a properly decoded WAV.
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level, espcially after the high-cut of the de-emphasis curve, so I
boosted the overall level almost 10 dB in addition to the de-emphasis
high-cut.
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with emphasis.
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is acceptable may change
over time. For this reason, I feel that one might be making a decision
they would later regret by using a lossy CODEC to encode their
collection.
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have not used ALAC, but I'm guessing that it would pass the
same sort of lossless tests since it is called lossless.
So, no, it does not make sense that FLAC and ALAC would sound
different.
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Timothy
can detect? Almost certainly.
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) kept it locked up big time so they can
make money from the licensing fees.
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was the
dbx-encoded LPs. Those blew me away at back in the day. No
inter-track noise on dbx LPs; the track would fade to complete silence.
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was
available.
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also go to the trouble to ensure the noise floor of their transfer
equipment is below the analog tape noise of the master and there are no
clipped peaks...
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be possible to extract to the hard drive totally
in the digital domain. FLAC, WMA, et al support multi-channel, and
Music Giants offers 5.1 downloads (although I'm not so sure how
pristine is the lineage of their HiRes files).
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be fun to try.
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precise than humam hearing Anyway, what does MW specify is the
improvement?
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As a matter of fact I used the FLAC Frontend yesterday to encode the
24/192/2.0 from the Classic HDAD of Alan Parsons' Eye in the Sky. It
can't do the ReplayGain calculation above 48 KHz, so that option must be
unchecked, but it encoded fine. Rather slowly, but fine.
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HalleysComet;267786 Wrote:
The analog stage in his Transporter modification is hardly extremely
standard
OK, what is non-standard about his analog stage? Does he use a
non-standard circuit topology? What is unique in his circuit?
--
Timothy Stockman
used signal transformers in a few strategic places when
I build audio equipment. Typically Jensen parts. But it has made me
aware of the advantages as well as the shortcomings of transformers.
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Timothy Stockman
Timothy
is because that's the way it was recorded!
My other beef is the mastering engineers who play mono tapes back on
stereo tape machines...
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of the web interface
where you click the starting letter then scroll to the album artist...
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be less traffic rumble.
Recordings can theoretically have a wider total window of dynamic range
if they are mixed so that the overall system gain at the studio is
adjusted as the recording progresses. But it is rare to see even much
above 60 dB S/N in a popular or jazz recording.
--
Timothy
jitter which an oscillator inside the
DAC can deliver to the DAC itself. This will reduce the jitter induced
sidebands, which arguably are potentially more audible.
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,
the audio is decoded, another MD5 hash is created and compared with the
hash generated when it was originally encoded. This proves, among other
things, version compatibility if it was encoded with one version but
decoded with another.
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DAC which can generate its own clock
by slaving to the DAC's clock.
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was played back to cut
the glass master, so such an error is embedded in the CD itself.
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-ripped from the source.
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to a tree which contains FLAC files or
a hierarcy of subdirectories with FLAC files.
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there, but they're hard to
use compared to EAC/FLAC, iTunes or WMP.
My recomendation: EAC/FLAC...
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to ensure that ite internally
generated message digest is correct.
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and the process
of ensuring data integrity on upwards of 1TB of material becomes
completely unmanagable. So, IMHO, the trick is determining what level
of human involvement gives one the best chances of keeping the data
intact over the long haul.
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Timothy Stockman
, and I may just see how many of my audio
tasks I can move to Linux.
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and string
quartet with her synth. I've been a big fan of hers since her first
album, Seven Waves and have many of her synth and piano albums in my
collection.
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The track Rain from the Suzanne Ciani disc Hotel Luna has some
fairly interesting stereo effects with the synth-generated raindrops.
I think the key to hearing it the way it was intended is to have good
phase linearity in your speakers.
--
Timothy Stockman
standard, but it is did produce quite convincing
phantom sounds, not just between the speakers but above, below and well
past the sides of the stereo speaker array.
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cases, Exact Audio Copy is able to extract
bit-accurate data from an audio CD.
Bottom line, a file properly extracted from a CD will be identical,
bit-for-bit, with the original file that was used to make that CD, so
merely recording audio on a CD makes no difference in the audio.
--
Timothy
correction and block addressing were added to bring the CD's data
reliability and repeatability up to the much higher level required for
computer data. Redbook CDs do not have this additional error
correction information, and thus it's harder to get a correct read from
a CD than a CDROM.
--
Timothy
I've already
done.) So likely, I'll make a third complete backup to be stored in my
self storage locker.
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the problem by making
a downsampled copy at 48 KHz, to which the encoder will add ReplayGain,
then I copy the ReplayGain tags to the original 96 KHz file. As a
bonus, I have a properly downsampled 48 KHz file for devices (which is
most of them) which can't handle 96 KHz.
--
Timothy Stockman
it was not a hard decision for
me to chose FLAC lossless. And another advantage of FLAC is that I'm
not at the mercy of Microsoft (WMA lossless) or Apple (ALAC).
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minutes.
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is used properly, the audio signal doesn't
spend a lot of time in the peak extension companding region.
Sorry for the topic creep.
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in
use for telephony, mu-Law and A-Law. Maybe a better question is why
did the CD standard pick 16-bit linear PCM instead of extending one of
the telephony standards.
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this isn't too confusing...
This becones more obvious if one ever has the chance to look at the
video from a (now obsolete) 1630 or F1 format digital processor.
http://en.wikipedia.org/wiki/Two's_complement
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Timothy Stockman
by the sample rate. Not
to mention audible and/or measurable effects of the filter.
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Hope this doesn't open up a can-o-worms, but isn't the Scott Nixon DAC
one of those non-oversampling ones with not much of an anti-alias
filter? I guess they count on the listener's ears doing the anti-alias
filtering... But what about the tweeters?
--
Timothy Stockman
it
personnally; the wosrt that ever happened to the Phase Linears in my PA
rig back in the day was that they overheated and blew the fuses. :)
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be the
bandwidth limit and how it should be enforced. How much do components
above 20 KHz matter to the average human? What are the tradeoffs
involved in filter slope as the Nyquist limit gets farther from what is
important to the average human?
--
Timothy Stockman
at a oddball sample rate.
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!
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Isn't 10/100BT already transformer isolated?
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samples, which can be
fairly inaudible.
So maybe what is being compared is normal mode (with no ASRC) to
broadcast mode (includes ASRC). I would think that the ASRC would
give Internet radio the potential of sounding worse,,,
--
Timothy Stockman
to be one master and everyone else is a
slave... My guess is that Slimserver is the master and the SB become
slaves in synchronized playback mode.
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Very interesting... I would have thought that it would be much
preferable to have an almost inaudible discontinuity by dropping or
duping an occasional sample to prevent a big rebuffering
discontinuity with buffer underrun or overrun...
--
Timothy Stockman
What??? You don't have expensive audiophile ethernet cables or an
audiophile router? Surely you must be getting substandard quality from
those Linn studio master files... :)
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Timothy Stockman's
audio!
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, making it possible to easily scan for HDCD audio
with an automated script. I agree with you, wrapper scripts offer
great flexibility.
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the vinyl if it sounds halfway decent.
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.
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attachments
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|
+---+
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.
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...
BTW -- Looks as if the term needledrop has migrated here from the
Steve Hoffman forums.
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you can use the word
clock on the Transporter to lock the digital source to the
Transporter's clock. Is this possible? Has anyone done it?)
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by a variety of existing devices, including
the Transporter.
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for high quality
audio distribution.
http://www.korg.com/gear/info.asp?A_PROD_NO=MR1000
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.)
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___
audiophiles
seanadams;215491 Wrote:
You realize it's converting the signal to PCM over s/pdif, right? It's
not a DSD output.
Yes, but at least its hi-res PCM. Would the Transporter know what to
do with a DSD file, anyway?
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Check this out: http://www.youtube.com/watch?v=xkkqsN69Jac
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the
transmitter. I resigned shortly after cranking up the processing.
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, well, word clock is the digital audio
equivalent that allows digital audio devices to run i sync.
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of metaflac commands (this is what the tag frontend does) to
add track numbers and ISRC codes extracted from CUE files to FLAC
files, fix/add the REPLAYGAIN tags and save the tags so that they can
be reapplied after a file is decoded to WAV and re-encoded to FLAC.
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