D
E V E L O P M E N
T O F
A H I - F I
This
is my
own project of a Non Over Sampling Digital to Analog Converter using
paralleled Philips TDA1543A, DIR9001 SPDIF receiver, passive I/V
conversion and
Direct Coupled output. I followed the advices of very experienced
people that I personaly know
and also used lots of information on the subject that are available
online. It all started
with Ryohei Kusunoki´s paper
on the MJ japanese magazine, which was
brought to my knowledge by my
friends
Marcelo Sfoggia (the famous recording and circuit's guru) and
Márcio Chiaramonte, to whom I thank very
much. I also thank
André de Bacco for information on the fabrication of PCBs with
laser printers and home iron, which by
the way motivated me to give a course to
my students showing the
technique. On the Internet, I
used most the information provided by Doede
Douma, schematics and ideas
from forums like
us.hifidiy.net,
www.diyaudio.com, www.diyhifi.org and many others. I also wish to thank
Elso Kwak, Luigi Carro, and other friends for personal communication
and comments about circuit details. Finally, I need to
acknowledge my
fiancé Rose
which not only accepts me working until 4am but also helps in some
steps of home fabrication.
by Marcelo Johann
As you may notice, I am writing the least I can, so this page
started
like a placeholder. The few sentences below include some details that
you are probably not
going to find elsewhere. Let´s tell the main part of the story by
pictures...
I included a better description of the project at the very end. Check
it out!
Early
stages:
DIR
9001 does work with TDA1543A in mode 01: 24-bit right aligned.
First
sounds comming out
Now
CS8414 with Mode 11: local clock drives everything but there is no
reclocking. In this mode, CS8414 sends a variable number of bit clocks
per sample
to keep synchronized to both SPDIF´s rate and local clock. You
see a
clock divider at right, hex inverter for input and clock buffering, and
a shift register at the
left to delay the data line by 1 bit transforming left-aligned mode to
I2S. This works only with TDA1543 (not Japanese A) I used the
exact 64*fs local clock, there is no reclocking, no synchronization
problems, but it does
not mean this structure is free from clocking problems, as the drift
between source and
local clocks will cause time distortions, perhaps less frequent.
First
SMD prototype being tested repeatedly due to a reset issue
From
the lab to the living room status! After many tweaks, I think this
summarizes how it sounds: hard to believe!
Here
you can see the 66MHz reclocking add-on board, disconnected.
I did not like
how it sounded in this design. I will retry it later on with better
components.
Here goes a
general introduction:
To listen to digital sound, a circuit must first convert a sequence of
numbers into an electronic signal, which will be amplified and
converted to air waves by a transducer. The SPDIF is a standard
employed by the majority of the audio equipment for the transmission of
sound data, and encodes both data and clock, being asynchronous.
Although there are ways of synchronizing a system by a second cable
that carries only clock, most consumer equipment has no such option,
and even in a professional studio there are situations with loops where
there is no way to select a single piece that has a high performance
clock close to it and slaves all other devices. So, most digital audio
circuits have to recover the reference clock from an external source,
and this is exactly the case for home digital audio reproduction. In
this scenario, we want to implement a separate Digital to Analog
Converter that synchronizes and listens to an SPDIF connection, so that
in the end we can generate a sound signal with much better quality
compared to what our digital source, like a regular
US$50 CD our DVD player, can deliver.
Circuit concepts and tricks:
DIR9001 is a modern SPDIF receiver by Texas Instruments that outputs
data with a clock jitter rated as low as 40ps RMS for 44.1KHz signals
with 512*fs main clock in its PLL. Clock jitter is a very important
subject for digital audio reproduction, as it distorts the reference
time domain, and its effects are very audible. To complicate it
further, the jitter may have different spectrum components, which will
translate into different performances on each part of the audio
spectrum. TDA1543 and TDA1543A are "economy versions" of a DAC by
Philips that are known to sound incredibly well compared to even the
best recent designs. The idea of Non Over-Sampling (NOS) comes from the
fact that to theoretically satisfy the error requirements in
time*amplitude an oversampled converter would have to present clock
jitter unatainable in actual scenarios, as first stated by Mr.
Kusunoki. To improve jitter, Kusunoki introduced asynchrounous
reclocking. In this scheme, a clock many times higher than the bit
clock (say, 50 or 60MHz) is used to re-clock the bit clock, slicing it.
This new clock absorbs all jitter within its period size, at the same
time introducing frequent shifts when the jitter or clock drift is
larger than it. Therefore, the result depends on the quality of the
incoming clock, the quality of the new high frequency clock, and also
on the interactions between both. The idea of NOS is also frequently
associated to the concept of filterless DAC. A filter is said to be
required after DA convertion to smooth out the stepped waves created by
the digital numbers. However, all filters degrade sound quality, as
they implement memory processes, with an innertial behavior depending
on the previuos samples. This smears out the information in the time
domain and decreases time integrity and correlation. On the other hand,
if we simply use the DAC with no output filter, the equipment that
follows it, most probably the speakers, and finally, the human auditory
system, will naturally filter out components above the audio spectrum.
Oversampling is used with digital filters and higher bit counts to just
implement the filtering, and that is why people mean filterless when
writing just NOS. After the DAC chip, a resistor is employed to convert
the TDA1543´s current output into a voltage, and this is supposed
to beat any other method, as it has the smallest number of components.
In its article, Mr. Kusunoki also introduced the idea of using
paralleled DACs to statistically reduce erros, among other reasons,
what is easily done with the output currents of the TDA1543. Since
those ideas first appeared, there has been an increasing number of
people implementing their own DACs both as hobby or as small commertial
projects. They sound incredibly well, clearly beating up commertial
products that are well known and used worldwide by audiophiles and
studios. In my design, I wanted also to get rid of the output coupling
capacitor, and for that I shifted the whole power supply down as to
bias the external ground reference to 3.85V of the original ground,
which is the output DC offset recommended by Doede Douma to minimize
THD when DACs operate at around 8V supply.
Resulting Sound:
As in most of the other reports, my implementation exhibits the
following characteristics:
- an extended sound stage, letting me perceive much better the 3D
placement of
acoustical recordings;
- reveals low level information that makes the sound become more
present, with a better sense of reality;
- better timbre: instruments sound more close to reality and different
from each other, more like in an analog recording;
- better bass and punch. It is surprising how much improvement we can
get in low frequencies. Part comes from PSU and DC connection,
naturally, but other subtle circuit tricks including clocking also
change LF a lot, as already reported by many.
Directions:
As I said earlier, I could not get an improved sound by using
reclocking, but it was an initial, quick implementation with average
components. It seems I got a boost in medium frequencies that made the
sound jump outside, but it definitely lost some of the improvements in
subtle distiction of instrument sounds and the clarity and extension of
bass. I also could not try eight TDA1543s in parallel with confidence.
The circuit is sitting open on the floor, may be the noise was already
at its
limits, so when I hooked up 8 chips, the DIR9001´s PLL very
hardly stayed in synch. Therefore, my next step is to redesign the
whole power supply. Afterall, Kusunoki´s published schematics had
already 90000uF in its PSU, with separate transformers, LRCRC filter,
and so... There is a thread in www.diyaudio.com entitled "Building the
ultimate
NOS DAC using TDA1541A" with over a hundred pages full of discussions,
new ideas and points we must consider for improvements, many of them
specifically for the TDA1543.
How I am listening:
I am using four Event 20/20 passive monitors together with a pair of
Jordan JX92S fullrange and a pair of NorthCreekMusic D2506S tweeters.
Everything is currently being driven by a brazilian Nashville Power 70
amplifier. I have also a Hafler Transnova P3000 (-3dB from 0.15Hz to
300kHz), but in the living room I prefer this Nashville for it is
smoother, with a response within -3dB from 5 to 150KHz, THD < 0.03%.
I know that the bass reflex studio monitors would not be able to
provide an accurate judgement for more subtle improvements or
decisions, mainly regarding bass, neither will allow my limited 4.5m
room, but that is what I got so far. When I talk about improvements, I
am comparing the outcome to a Metric Halo ULN-2 firewire Interface
which is a studio equipment, and is in the same league as an Apogee or
RME converter, if not better, which are all in the range of US$1k to
US$2k. So, although I mentioned US$50 players at
the begining, this is not the kind of equipment I am comparing against.
In fact, I cannot even recall when was the last time I listened to the
terrible sound that comes out from these cheap consumer players. It is
also worth to highlight that the devices to which I and others are
comparing are 20bit, 24bit, 4-16 times oversampling, and it is
interesting we can make a 16-bit NOS DAC that sounds better. So,
caution to not be fooled here again by the number's game. We still
listen to 44.1kHz, 16bit CDs, and even at the studios it is difficult
to make a better sound out from equipments that exhibit higher numbers.
Marcelo Johann,
2009
johann at inf ufrgs br