Thermionic Valve Analogue Stages for Digital Audio

A short overview of the Subject by Thorsten Loesch

1. Introduction

In the last decade the whole issues how to interface the Silicon nastiness of
CD-Players and Digital to Analogue Converters to Valve Equipment has received
extensive coverage in the Audiophile and DIY Audio Press. Much polemic has
issued forth. A number of companies have even introduced add on Valve Stages
claiming them to be "the missing links" and the like. This brief article shall
not concern itself with the polemic surrounding the issue, or with specific
criticism of many implemented solutions.

Nor shall we discuss the inherent futility of attempting to "reform" or
"improve" by introducing Valves, a Digital Audio Format which not only suffers
from enough build in limitation as to make true "High Fidelity" inherently
impossible and for which the vast majority of recordings not even make use of
the available quality. Nor shall the issues of massively incompetent digital
design (clock jitter, PSU decoupling, managing of RF Noise and so on) in almost
all commercially available equipment be discussed here in length.

Before continuing with this Article the august reader is advised to make
herself or himself comfortable with the underlying principles of Digital Audio
and CD Player in special as no further consideration will be given to this
topic.

Instead we shall concern ourselves only with delivering the however
compromised output Signal from the Digital to Analogue Converter to the Output
Jacks with the lest amount of further perceived degradation.

Considering this task, the use of Zero Feedback circuits seems almost
mandatory. This is because with a DAC or CD Player we deal with a generator of
massive levels of wideband RF Noise which, if used with conventional feedback
based circuits will lead to severe levels of intermodulation distortion. This is
easily observable with noise-load testing as much as 1% with leading High End
Converters and worse for much consumer equipment.

So, having defined the almost absolute need for zero feedback circuits if our
goal is to offer high fidelity without recourse to repeated design iterations
and use of expensive superfast, high resolution FFT Analysers and noise-load
testing, the only readily available devices with an ability to offer low enough
distortion levels in zero feedback mode are thermionic valves, most notabene
triodes.

Now, how do we interface triode valve circuits to the kind of DAC Chip’s
commonly found in consumer digital equipment? To answer this we must first
understand that currently two fundamentally different DAC Architectures are
being used and that the two different Architectures require a fundamentally
different approach to analogue stage design. The two DAC architectures in
question are multibit and timeslicing DAC’s. Let’s look at the technology.

2. Multibit Digital To Analogue Converters

The Multibit DAC is the classic technology used from the beginning and has a
very long history outside digital audio. In most simple terms a Multibit DAC
contains a number of switches and a resistor network (and some auxiliary
circuitry). Each of the switches corresponds to a "Bit" and will switch in a
resistor allowing a certain current to be presented on the output of the DAC
Chip.

If take the classic Philips TDA 1541 Multibit DAC as example, the MSB (most
significant Bit) will toggle on or off a Switch which will switch on or off a
current of 2mA. With all Bit’s at Zero all switches will be off and the output
will be at zero. Switching the MSB on will produce -2mA current and represents
basically "digital silence". Switching all other 15 Switches on will add another
-2mA to the total now bringing the output to -4 mA. So switching from all bit’s
off to all bits will change the Output Current from 0mA to -4mA. Many more
modern DAC's use a seperate offset current source (often unaccessible for the
user of the chip) to bring the output at "digital silence" to 0mA, giving a
+/-...mA Modulation.

As this current is normally derived from a reference voltage via precision
trimmed resistors the output needs to be held close to zero volts in order to
avoid introducing distortion. However, it might be noted that the distortion
actually introduced is of a type similar to that of a triode and hence, if we do
not specifically concern ourselves with measured performance we can more or less
ignore this requirement within reason.

However, regardless how specifically we convert the output current of the DAC
into a voltage (in the easiest way simply by using a small value Vishay bulkfoil
resistor), with a classic Multibit DAC we generally require some form of voltage
gain and quite a bit of it.

Commonly used is a op-amp chip where the output current of the DAC is applied
to the inverting input of the Op-Amp, which due to it’s normal function presents
a virtual ground. Due to the steep slopes of the steps in the output current
between changes in the bit’s we require a device in this position that is not
subject to limitation by it’s Slew-Rate. If this is not observed severe
Transient Intermodulation Distortion (TIM) will occur which while very audible
as aggressiveness, clipped sibilants and similar artifacts that are relatively
hard to measure, using conventional methods.

In many existing Players and DAC’s the used Op-Amp’s are NE5534 or NE5532 or
functionally equivalent types from JRC/NJR. All these devices are incredibly ill
suited for this purpose and no doubt the often less than ideal sonical character
of early CD-Players can be blamed at these devices. In the practical section we
will look at options for doing all this with Valves and somewhat better.
However, the coverage of classic Multibit DAC’s is somewhat redundant, as they
have been almost entirely replaced with the other type of DAC, the one I refer
to as TIMESLICING DAC.

Multibit DAC’s where (and are) made chiefly by Burr Brown (PCM56, 63, 67, 69
and PCM 1700,1702, 1704) Analogue Devices (AD1862, AD1865) and Philips (TDA1541,
1543 and 1545). Others exist and any of the above Manufacturers also made custom
chips for large OEM Manufacturers like Technics/Panasonic, Sony and so on). The
classic Multibit DAC’s often require fairly complex PSU Arrangements and are
comparably expensive to make.

In the extreme High-End Equipment (Spectral, Mark Levinson, Krell, Audio
Synthesis and Wadia) Multibit DAC’s are still the order of the day. Multibit DAC
Holdouts in the better budget consumer CD-Player Market are Rotel and Denon. All
but their really bottom feeding range uses Multibit DAC’s. The top of the line
Teak and Pioneer CD-Players (Pioneer only in their latest Model) also use
Multibit DAC’s. Linn and Naim have been long-standing advocates of Multibit
Chips though their more recent lower priced Models no longer use Multibit
DAC’s.

Almost all pre 1990 CD-Players are of the Multibit Type, however, before
trying to hunt such a device down to modify it consider that they have seen a
lot of service in the last 10 to 16 Years, spares tend to be unavailable and
breakdowns are likely.

One possible option would be to buy (very cheaply) one of the earlier Philips
or Marantz CD-Players and remove the SAA7220 digital filter and the TDA1541 DAC.
From these one can make a stand-alone DAC, similar to the one offered by Curcio
Audio (from where one could obtain suitable PCB’s) or the van Alstine Valve
DAC.

With budget standalone DAC’s one tends not to find Multibit DAC’s at all, as
they became common only in the 90’s. And to buy an older "High-End" DAC for
modification is often also not such a good Idea as their input circuitry is
often jitter ridden and hence will not allow much scope for sound quality.

The best bet perhaps is to start from a Denon (the 825/835 Models and
upwards) CDP, a suitable Rotel CDP or the Pioneer PD-S06. Otherwise, the above
section is primarily history, for in the current days reigns supreme the
Timeslicing DAC.

3. Timeslicing (or Delta Sigma) DAC

The device I call Timeslicing DAC is known by many Names. Technics calls it
"MASH", Pioneer calls it "Pulseflow" and Philips calls it Bitstream and so on.
These DAC’s are often also referred to as Low-Bit or single DAC’s. It is in many
ways an ingenious Device.

Instead of attempting to make precision analogue silicon (always expensive),
they circumvent the issue by using only one "on/off" switch and by toggling this
switch very, very fast. The result is a squarewave of very high frequency which
is however symmetrical. So once we pass this squarewave through a lowpass filter
we are left (assuming Digital Silence) with a voltage of zero.

If we now start taking, let’s say, every 65536’s pulse out we will achieve a
tiny deflection of the Voltage. Depending if we omit the positive going pulse or
the negative going pulse we will get a deflection to the positive or negative
side.

Now for this to really work for CD Style Digital Audio we theoretically
require 216 Pulses (65536) pulses for every 44.1kHz Sample. That
means we need theoretically a 2890 MHz (or about 2.9 GHz) Clock to run the DAC.
This is clearly not practical and a number of mathematical "tricks"
(noiseshaping especially) are employed to get a reasonable performance at much
lower Clock Rates (around 16.9 or 33.8 MHz).

Usually the DAC Chip switches the PSU Voltage (+5V) and in order to make the
output more immune from the PSU noise all Timeslicing DAC’s use a differential
output. This means we normally have a positive and a negative output for each
channel. These outputs are voltage based and usually carry a 2.5V DC offset and
have a swing each of +/-1.4V peak or 1V RMS. This enough voltage to require no
further amplification.

These two outputs however need to be combined into a single-ended output and
then further lowpass filtering is needed as the signal is overlaid with large
quantities of ultrasonic noise. This is actually much more problematic than the
already problematic situation with Multibit DAC’s. As a result many designers
screwed up their analogue stages for the early Timeslicing DAC’s up so badly
that most DAC Manufacturers started integrating various parts of the analogue
stage onto the chip to avoid this (notable exception - Nippon Precision Circuits
- NPC).

Later Phillips Bitstream Chip’s (like TDA1547) are famous for having been
crippled sonically by some Idiot who integrated a analogue stage based on the
NE5532 Op-Amp onto the silicon. Another DAC vendor who really made a sonic mess
is Cirrus Logic (under the Crystal Brand). After their superb CS4303, which
offered pure digital outputs (and which almost no-one got to sound good as the
analogue stages could not cope with all that RF Noise) and the CS4328 which
implemented an excellent internal analogue stage, all their other DAC’s (CS4327,
4329, 4390 and so on) have incorporated switched capacitor lowpass filters
(sonic poison) and lousy CMOS Op-Amp output buffers.

While these measures make the Crystal chip so easy to implement that even the
most incompetent and hamfisted designer cannot screw up the sound of the chip
too much, they seriously massacre the inherent sound of the device. This makes
(all other things being equal and a competently designed analogue stage assumed)
the current generation of Crystal DAC’s by far the worst sounding Timeslicing
DAC’s yet, closely followed by the latest generation of low-cost DAC’s from Burr
Brown.

A number of recent Timeslicing DAC have gone further and even incorporated
the rest of the analogue stage onto the chip (Philips TDA1305 and Burr Brown PCM
1710/16/28/32 come to mind). Just as it the habit of large corporations, these
Op-Amp’s used on chip are not your audiophile favorites. On the contrary, with
Philips it is of course the equivalent of a NE5532. Burr Brown did not even
spring for something like their well-respected OPA2604 or OPA2132/2134, they
used a really nasty CMOS Op-Amp.

Of course, these Op-Amp’s you cannot see. Nor can you bypass them anymore.
They are always there and inside the Black Box you think of as "DAC".

But back to basics. The big advantage of the Timeslicing DAC is that it is an
almost pure digital circuit. All and everything can be done in the digital
domain, so much so the output from earlier Timeslicing DAC’s is a pure
pulsewidth modulated squarewave. And unlike precision analogue Chip’s, even very
complex digital chip’s can be made dirt-cheap. Even the integrated analogue
circuits are usually of the "5 pence the piece if you buy 1000" type, making not
much of an impact on production cost. And let’s not forget that many Multibit
DAC’s require four to six separate supply-voltages. The Timeslicing DAC usually
requires only one and that of course costs a lot less again.

And the production cost is exactly the reason for the introduction of
Timeslicing DAC’s. Any claims for superior performance or any better technology
and such, can be safely relegated into the realm of fairytales spun by marketing
futzies. Neither sonically or measurable are there ANY advantages for the
current Timeslicing DAC’s over classic Multibit Designs. On the contrary, they
are technologically much less capable of high fidelity reproduction than
Multibit DAC’s and as they operate on the basis of clock cycle modulation any
instabilities of the master clock (jitter) have a much larger impact on sound as
with the old Multibit Designs.

However, the presence or absence of Multibit DAC’s or Timeslicing DAC’s
should not be seen as an absolute indicator of quality. As usual, much depends
upon the implementation. This is shown by the fact that several Digital
Processors or CD-Players using Timeslicing DAC’s sound very good (Timbre DAC,
Acoustic Precision EIKOS CD Player) and certainly a lot better than the worst
Multibit DAC sporting Machines.

I did feel that the discussion of the different technologies was necessary,
as too often even experienced Designers tend to view DAC’s as a "Black Box" with
little concern about how they really work. Together with mistaken assumptions,
based on the datasheet of the DAC issued primarily not as design-guide but as
thinly disguised marketing material, one often sees a lot of effort and care
being lavished on designs based on Chip’s that should be buried quietly in a
dark spot around midnight and not used in quality audio equipment.

With all this bleak view I have of the CD digital format, it’s recordings and
the competence of the designers of much of the equipment used to play CD’s one
is tempted to ask: "Why bother?". And I must say that I have almost taken this
view. With a huge (and still fast growing) record collection and excellent
analogue replay equipment I personally can do (and mostly do) without CD’s.

However even I as founder (and currently only Member) of "The Holy Order of
the Latter Day Luddites" own a CD-Player (a DVD Player actually) and own a
number of CD’s. Indeed, as I type this I’m listening to Phil Collins from CD.
Others have committed themselves entirety to the CD Medium. I doubt that ANY
current audiophile or ultra-fi Audio-System is without a CD Player.

Love them, hate them, we have to make do with them and then extracting the
last bit of quality becomes essential.

4. Analogue Stage Requirements

After having discussed the different types of DAC we can now sit down and
formulate a few basic requirements for analogue stages (solid state or
thermionic valve based) to be used with the respective DAC architectures.

An analogue stage for current output (Multibit) DAC’s requires a low input
impedance (the lower the better - about 100 to 200 Ohm appears to be a sensible
maximum), a fairly high voltage gain and while it must withstand some notable RF
Noise, no extreme lowpass filtering is required.

This suggests a resistor for I/V conversion, followed by a suitable gainstage
with plenty of gain, to present a reasonable output Signal. Schemes for this
have been long available and what is presented below for Multibit converters is
mostly a rehash of other peoples work with the odd little touch from me here and
there.

An analogue stage for timeslicing (Bitstream/Delta Sigma) DAC’s requires a
very low voltage gain, notable RF Filtering to protect following equipment from
high levels of ultrasonic noise. On the surface it also seems to require a form
of balanced to single-ended conversion. It is the latter requirement that has
thrown many people off track.

I have seen one solution (by Chris Found) which uses differential SRPP valve
stages but ends up with by far too much gain, excessive output voltage and a
substantially unnecessary complexity. Another rather ingenious solution (Sephano
Perugini) was to use a transformer, which of course works great, offers galvanic
isolation form all the noise in the DAC and allows bandwidth limiting (to
eliminate ultrasonic noise) in the most simple and direct way.

While each solution has it’s merits, one is too complex and has too many
glowing bottles while the other is really ingenious but has no glowing bottles
and relies upon a very high specification 40k : 40k Transformer, neither a
readily nor cheaply available part. A pair of good quality, wideband 40k
line-level transformer can easily exceed £ 60 and the best from Jensen cost
several hundred dollars each.

So I rejected either approach and looked again at the issue. I decided to try
the Microsoft approach* and decided to "just keep driving because no-one will
notice anyway". So the output was taken from only side of the balanced pair.
Some form of lowpass filtering is needed together with a buffer stage. An LC
Filter followed by a cathode follower seems to answer the purpose admirably. My
own original solution omitted the cathode follower entirely and later introduced
a J-Fet based buffer. This works and sounds very good indeed.

All this said, under certain conditions (amplifiers or preamplifiers with an
input impedance of 47kOhm or higher) we can completely omit ANY analogue stage
for Timeslicing DAC’s and use only a simple passive circuit, avoiding any
additional active components. This is especially true for recent devices from
Burr Brown (PCM 1710,1716,1728,1732 Series), Analogue Devices (insert list of
devices) and Philips (TDA1305).

Also for Multibit DAC’s it is possible to completely avoid active analogue
stages if the following amplification chain has enough spare gain available
(usually one would expect to see about 25 to 100mV RMS Input sensitivity for
clipping).

The cases where one can afford to operate without external analogue stage are
comparably rare though. I might discuss the "DAC-Direct" Output Topology I have
developed for both types of DAC’s another time. For the time being, lets get
practical.

*Three Microsoft VP’s, notably a Marketing Manger, a Helpline Manager and
a Software Engineer drive down the Autobahn in a BMW. The Car has a flat Tire.
The marketing futzi immediately proclaims "We need a new Car!". The helpline guy
say: "Just a minute, let me ring up the BMW Helpline, maybe they can ship us a
new Wheel?". Finally the Software Engineer says: "What’s all the fuss about,
guy’s? Just get in and keep on driving, no-one will notice!".

Practical Implementations

1. I’ve got the Power - a powersupply digression

Prior to launching into a lengthy discussion of the specific example circuits
for analogue stages let’s briefly cover a simple, fairly inexpensive but high
performance powersupply for these circuits. For simplicity of procurement it
primarily uses components from Maplin.

Maplin have an inexpensive Valve Mains Transformer in their program, as well
as inexpensive but adequate chokes. Used for this PSU is one XP27E Mains
Transformer (around £14.00) and two ST28F Filter Chokes (around £ 7.00 each).
Other sources exist, but I seriously doubt that a 50mA 20H Choke and a suitable
mains Transformer can be obtained for under £ 30.00 the set from any other
source. The quality is more than adequate.

Most other components are standard industrial quality ones. The electrolytic
capacitors in the HT Supply should be of high quality but it is not necessary to
spring for Elna Cerafine’s or Black Gate’s. The lowest cost suggested devices
are Nichicon VX Series axial types, available from good electronic
stockists.

The Circuit for the Supply is given below.

I believe one or two unusual features in this supply need explaining. First
looking at the HT Supply there are a number of unusual networks around the very
standard rectifier bridge. These are snagged from the late John "Buddha" Camille
and have two purposes. First of all the CRC pi-networks ahead and after the
rectifier bridge help to reduce any noise. The use of carbon composite resistors
in this position is mandatory to avoid resonance effects.

However, the chosen resistor values are unusually high, for a reason. They
simulate the anode impedance of an EZ80 in full-wave mode. The resulting
rectifier (with added networks) behaves very much like a valve rectifier,
removing the often-notable unpleasant sharpness when solid state rectifiers are
used. The diodes used should be soft recovery types but are relatively
uncritical. Without having tested all available diodes I recommend the
Telefunken BYV and BYW Series as well as the Motorola MUR4XX Series. All these
cost less and sound as good as the much-vaunted "Hexfreds".

Furthermore, the HT Supply circuit operates in choke input mode. For load
currents of about 12 to 15mA the Maplin chokes can be used for choke input
supplies. It is however necessary to place a piece of rubber between chassis and
core to securely and strongly clamp the choke’s core in place as otherwise the
core will buzz. I have myself used these chokes for up to 40mA choke input
supplies in the described manner and fashion with no ill effects. Using one
choke in each "leg" offers improved common mode rejection and doubles the
Inductance.

The rest of the circuit is bog standard, though the noise fanatics in the
audience might want to replace the 1k filter resistor preceding the 220uF main
filter capacitor with another Maplin choke. In my books the 1K filter resistor
is just fine. Not shown here is the final filter stage that for each individual
channel will be shown in the schematic for the analogue stage. In any case these
will be a 22uF Ansar Supersound (or Solen) polypropylene capacitor combined with
a 4k7 Resistor.

It should be noted that the above PSU Design is hardly ultimate or in any way
maxed out. Significant improvements can be applied. However, the deciding
criteria’s here where simplicity and reasonably low cost to get our project
underway. More elaborate designs are open to anyone caring to implement them.
Lastly, the very same supply is obviously suitable for any Circuit needing about
210V +B with 12mA (or less) current-draw and 6.3V Heaters with 0.6 to 0.7A
current-draw. For the heaters about 1.2A can be accommodated if one of the 0.1
Ohm Resistors and one of the 0.68 Ohm resistors is bridged out.

2. Thermionic Valve Analogue Circuits for Multibit Digital to Analogue
Converters

The circuits discussed in the following are suitable generally for any DAC
Chip having around +/- 1mA output current. This includes Analogue Devices AD1862
and AD1865 as well as Burr Brown PCM56, PCM67, PCM69, PCM1700, PCM1702 and
PCM1704. The Burr Brown PCM63 and Philips TDA1541 will require somewhat
different circuit values.

Due to a number of factors the circuit proposed is based around the SRPP
circuit. This tends to be somewhat colored sonically, however the specific
coloration’s seem to blend well with digital audio’s failings, producing a
generally subjectively pleasing sound in this context. In order to allow easy
procurements of valves and reasonable gain as well as reasonably low output
impedance the circuit is proposed for use with the ECC88/6922.

However, a 6CG7/6FQ7 can be used without changes and even a 6SN7 can be
employed if the higher heater current and the 8-Pin socket is accommodated. I
know we have a few people in the circle who think the 6SN7 a good sounding and
useful valve, even if I don’t. If desired the ECC83 or ECC82 can be used. In
this case the cathode resistors should be increased to 3.3kOhm.

Depending upon Valve used the output voltage of the circuit will range from a
nominal 2V RMS for digital full scale with the 6CG7, ECC82/12AU7 and 6SN7 in
circuit, 3V RMS if the ECC88/6DJ8/6922 is employed and 8V RMS if the ECC83 is
used. As a special case, the use of the 12AY7/6072A will result in about 3.5V
RMS out.

The circuit includes a lowpass filter and (I believe uniquely - at least I’m
not aware of any other design that does) the deemphasis correction. The lowpass
will produce a mild (0.7db) attenuation at 20kHz, which should be innocuous.
CD’s using deemphasis are rare, but they exist. Many older CD-Players and DAC’s
use analogue circuitry to carry out deemphasis, so I believe the function must
be provided where needed in the analogue stage. In this case R2 and C2 provide
it together with the switch. The switch is ideally a high quality relay (mercury
wetted reed relays are best) and needs to be controlled from the Players/DAC’s
internal logic.

The two parallel output capacitors are used to achieve a suitable load
compatibility. The capacitors used are very high quality industrial Foil &
Film types from Arcotronics or LCR and sold by respectively RS Components or
Farnell in multiples of 5 and for a sane price. Their performance is broadly in
line with Hoveland Musicaps. The resulting 0.94uF output coupling capacitance
will allow loads of 20kOhm or higher to be driven without audible bass rolloff.

While it is possible to further increase the capacitance, I would not
recommend to do it in order to driver lower load-impedance’s. For ECC88, 6FQ7
and 6SN7 the 20kOhm load possible in this configuration are the lower limit, for
ECC82, ECC83 and 12AY7/6072 the load impedance should not fall below about
50kOhm and hence in this case even a single 0.47uF coupling capacitor will
suffice.

All of this gives a basic circuit now almost a decade old and well worn, but
it still delivers the good’s. Here is what it looks like:

For anyone looking for something a little freakier, I can suggest the use of
the Euridice preamp circuit presented here in issue 34. Simply drop the Euridice
circuit in substituting the lower ECC88 Grid and loosing the rest. This will
then offer balanced or single-ended outputs, a very low output impedance and the
ability to operate with pro-audio 600 Ohm loads. It will also cost a lot
more.

Now to the PCM63 and TDA1541. The TDA 1541 is still one of the absolute best
sounding DAC Chip’s ever made (followed closely by the PCM63). Both Chips are
also fundamentally different from most other DAC’s of their kind by delivering 0
to 4mA output current. In order to keep the error due to the non-zero impedance
of the I/V conversion node limited it is essential to reduce the resistors in
the circuit accordingly (halve their values) and to increase the capacitors
(double their value). Yet we are still not there.

The PCM 63 is unique in that it allows us to disconnect the output from the
pull-down current source. No such thing exists in the TDA1541. In order to offer
a 0mA output at digital silence (that is NOT all bit’s set to zero by the way)
almost all Multibit DAC’s use a pull-down current source offering a - ½ full
scale current offset, zeroing the output at digital silence.

The PCM63 allows us to eliminate this current-source (a bunch of transistors,
resistors and Jah knows what else), simply by disconnecting pin 5 (offset) form
pin 6 (output). Having done that we are left with a DAC that will swing now from
2mA for digital silence to 0 and 4mA peaks for negative or positive full-scale
signals, respectively. It is in this mode that the highest degree of sound
quality from the PCM63 is realised, at the penalty of a ½ full-scale offset. In
many circuits the measures required to eliminate this offset would be worse than
the problem, so this is rarely done. In our own little circuits the offset
matters little and hence we can safely employ this scheme. The highest value of
I/V conversion resistor usable before significant distortion becomes notable is
100 Ohm in this case for the PCM63. Similar Limitations apply to the TDA1541

We hence need to substitute the following values:

R1 = 100 Ohm

R2 = 43 Ohm (44 Ohm accurately)

C1 = 33nF

C2 = 330nF (320nF accurately)

By the way, I’m looking at making a DAC using the PCM63 kindly donated by
Geoff Mead which will use the above with a Euridice style amplification stage
and NO digital filter. (added later - this developed into the TDA1541 based
"Adagio" DAC found elsewhere on this website)

The circuitwise similar Audio-Note UK made DAC5 (at a whopping £18,500
retail) was just noted by Martin Colloms in HiFi-News as "best from CD so far"
and uses actually a much inferior chip, the AD1865 a dual 18-bit Multibit DAC.
Martin noted a subjective quality approaching that of experimental new
high-resolution digital systems (96kHz/24bit/SACD). Perhaps it’s time to go out
and buy the last remaining PCM63 and TDA1541 before they walk in order to make
THE DEFINITIVE CD format DAC.

3. Thermionic Valve Analogue Circuits for Timeslicing Digital to Analogue
Converters

As noted before, the timeslicing or Delta Sigma DAC doesn’t really need any
analogue stage in order to provide enough voltage output. However, due to it’s
ill conceived modus operandi we require lowpass filtering and (if we insist on
being boring and conservative) a balanced to SE converter in case of most high
performance devices.

Let’s first split the range of timeslicing DAC’s into three groups. The
latest, most recently introduced group uses extensive on-chip analogue stages,
sounds by far worst of the lot and for some reason receives in it’s
implementations consistently rave reviews*. These are as mentioned the Burr
Brown PCM1710/16/28/32, Philips TDA1305, Cirrus Logic CS4327 and Analogue
Devices (insert list).

I suggest taking the hint from the manufacturer. Just take the output pin of
the DAC Chip directly to the output jack with only a good quality coupling
capacitor in-between. Believe me, it truly does sound best. The on-chip
filtering is usually sufficiently good to not cause any problems even with solid
state equipment. A valve buffer (cathode follower) can be employed, just as per
the circuit below, but again, I do not feel, that it improves anything.

I must also note that I personally find the sound coming from these new
devices to be exceedingly poor, when compared to really competently implemented
Multibit DAC’s or earlier generation Timeslicing DAC’s.

The generation before the latest designs is a bit more heterogeneous. We find
many DAC’s with differential output but various different schemes of on chip
filtering and buffering and the occasional DAC with a single-ended output and
really good on chip analogue silicon. In the latter case (CS4328, certain OEM
Products) the same as above holds true. For all other differential output DAC’s
the circuit shown further down should/could be used and should work under any
conditions, offering good filtering and ensuring an easy load for the DAC Chip,
keeping distortion low.

Finally, the original first-generation timeslicing DAC’s have no analogue
stages. Some of these are early Philips chip’s as used in the QED "Digit"
external DAC, also the NPC SM5872 (Marantz CD-53/57/63/67 among others) and NPC
SM5864 (Arcam Alpha - lower Models) and surprisingly many OEM chips even in
current use. All these will require some buffering and due to the lack of on
chip filtering a direct connection without any additional lowpass filtering a
direct connection even to high impedance inputs is not advisable. Again, the
circuit shown below will work just fine.

The circuit seems very novel, but it is not at all so. The differential
filter at the input of the circuit is used in order to present an identical load
to either output of the DAC and to implement a well-damped 3rd order
lowpass filter. Again a small loss (about 0.8db) is allowed at 20kHz. The
positive output of this filter is then buffered by a ECC88 cathode follower.

This cathode follower has been subjected to some improvements over the
classic version. By bootstrapping (or cascoding) the anode of the follower valve
the circuit is linearised. The use of a J-Fet connected as current source in the
cathode of the follower valve serves the same purpose. More on this circuit can
be found in Allen Wrights "Tube Preamp Cook Book" and hence it will be not
discussed much further.

The usual 2.5V DC offset on the DAC Output is put to good use in order to
provide additional bias to the Grid of the follower valve. This allows about 4
to 5V to develop on the cathode, more than enough voltage headroom to allow the
about 1V RMS output to be buffered with low measured (and perceived)
distortion.

It is possible to use a simple "classic" cathode follower in the above
circuit. This will certainly result in a much more "tubey" and possibly
subjectively preferable sound. In this case the ECC82, 6CG7/6FQ7 or 6SN7 seem
ideally suited with a 2.2kOhm resistor in the cathode. An ECC83 might be
employed with a 3.3kOhm cathode resistor.

The sound will be very different to the circuit shown above. Mostly
transparency will be substantially compromised, however the subjective result
might be considered pleasing. Perhaps it is worthwhile trying designs, the
classic and the "Super Linear Cathode Follower".

*Either reviewers are blinded (or deafened) by the "24-Bit" claims for those
Chip’s or the fact that finally the last bit of control over analogue
performance has been taken away from the CD-Player Designer stop’s them from
screwing up the Sound of potentially better sounding devices up. Whichever way,
I’m just not getting it.

Conclusion

Armed with the designs shown above and the background information presented
it should be possible to convert any available digital audio replay device to a
state with a more transparent, less subjectively compromised design. Most
CD-Players and DAC’s will have sufficient internal space to fit both the valve
circuit and the PSU parts.

The perhaps best mechanical solution for most CD-Players would be to obtain a
suitable sized aluminum box/chassis from Maplin (the 150mm X 114mm X 76mm one
looks right). This would be attached at the rear of the Player or DAC and one
would run some flying leads from the Player/DAC circuitry trough some holes in
the rear panel (how about the ones previously containing the output jacks) into
the valve stage.

All other components and the PSU could be placed safely in this separate
chassis and again, the mains power could be send through another suitably
drilled hole, making the whole solution neat, tidy and professional looking. If
done well enough it would even allow the Player/DAC to be returned to stock
condition with no major cosmetic blemishes.

Now I suspect that some people would think it having been convenient had I
listed in detail the relevant Pin’s on all noted DAC Chip’s and provided more
practical notes on the way these are accessed. This was not done for a number of
reasons. Primarily I feel that anyone should get a service manual for their
piece of equipment before starting to work on it. This will contain a schematic
that has the relevant information on it. If such a schematic, even if present
would not be able to convey the needed information, perhaps it would be useful
to take someone reasonably experienced with digital audio circuits and their
modification as "mentor" for the specific project (please DON'T bother e-mailing
me - I don't have time for that kind of mentoring.... sorry).