Subject: Class D OEM product FAQ
From: "Saved by Internet Explorer 11"
Date: Mon, 20 Jan 2014 09:55:34 -0800

Class
D Product FAQ

Dear
Reader,

Here's version "0" of a
class D product FAQ file. It was not my intention to give any DIY
information on building amps (that will be another document) but to give
some background information on the OEM class D market, for use by
people who are looking to improve their audio power products without
having to start their own research, as well as for plainly
interested folks (like myself). Someone had to stand up and
distill the info from the noise. I hope you'll find it
useful!

Tip Voigt ([email-masked])

What are the
reasons to move to class D ?

Class D amplifiers are
chiefly chosen for their power efficiency and all subsequent
advantages:
small physical volume, low weight, low air-flow
requirements, lessened power-supply requirements. All these properties add
up to vastly expand the product concept and design freedom. In a few
cases the high-tech appeal of a class D amplifier (then called a
"digital amplifier") is a deciding factor. Also, some class D
offerings exhibit sound quality otherwise attainable only with much
more expensive means.
Contrary to popular belief, class D
amplifiers should not be chosen for cost reasons. The most
cost-effective class D amplifiers only come close to the price of
the most cost-effective linear amplifiers. This is mainly because
heatsinks are not as expensive as they look.

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What kind of
amplifiers should be called "class D"?

In the light of the varying
terminology and trade names concerning class D amplifiers, here is the
best working definition of "a class D power amplifier" that I've
come across:

"A power amplifier
operating with all its power stage elements alternating between on and off
states"

Note that this definition
does not specify actual power stage topologies or modulation methods. It
also explicitly excludes improved-efficiency amplifiers in which a
linear power stage is embedded (Sunfire, Indigo BASH, Himmelstoss et
al, ...). Through their relative obiquity they are known as "poor
man's class D" not for financial reasons but because they are
invariably proposed by people who are unable to build a good class D
amp.

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Is a class D
amplifier a digital amplifier? When does it qualify as
one?

There is quite a tradition of
attributing the word "digital" to any class D amplifier. This is
because of the fact that the output devices in a class D amplifier
are principally operated in the ON and OFF states. However:

they spend between 1% and
10% of their time transitioning between those two states. What
happens in this transition interval has a defining impact on the
amplifier's performance. These errors are called timing
errors and constitute very analogue behaviour.

the amplitude of the
switched voltage is not fixed. Power supplies have a finite output
impedance and tend to have ripple on them. Class D power stages per
se have no power supply ripple rejection. Switch-induced ringing
on the power supply and power stage similarly cause amplitude
errors.

These
two error categories often make the intuitively apparent (and often
quoted) linearity of the output stage worse than that of properly
designed linear power stages, which can be very linear indeed. If
half-decent performance is required from even a well-designed power
stage it will inevitably require error control, mainly feedback in
its many forms. This has to encompass monitoring the actual
non-ideal power stage waveform, making it an inherently analogue process.

A good definition of a
digital power amplifier would be:

"A class D power amplifier
where:

the switching pattern
is a bit pattern

generated entirely
using digital means, independently of the output
stage

that is in its own
right a high-quality digital representation of the wanted
signal

the power stage is
designed with inherently minimal timing and amplitude errors such as
to directly offer high-quality reproduction of the drive
signal" (i.e. exhibiting minimal analogue
nonidealities!)

Conversely, an amplifier is
not digital if any of the following is true:

analogue processing is
present

the amplifier is not class
D

the input signal is
analogue

feedback or other error
correction is employed

large timing and/or
amplitude errors are present

It is interesting to find
that only one commercially available product (Tact Millennium) and none
of the OEM offerings qualify by this definition. Those designs
operating in open-loop with digital PWM but with normally performing
power stages (hence offering poor performance) will be called
digital-PWM amplifiers.

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What are good
reasons to choose for/against digital-PWM amplifiers?

For:

A controller built into an
already used DSP with only power components to be added is the
cheapest way possible to achieve a power signal reminescent of
an audio signal. If the controller is a separate chip (equalling
extra component cost), as is currently the case with all known
offerings, this argument becomes void.

Considerable tech appeal
of the words "digital amplifier" on the box. As said earlier, most
companies quite unashamedly use this nomenclature for perfectly
analogue amplifiers as well.

Against:

Obtaining acceptable
performance (THD<0.1%, hardly enough to classify as a digital
amplifier) is complicated and expensive. It requires an enormously
well-regulated power supply and a blazingly fast switching power
stage (expensive electronics, hard to tame EMI problems in the
FM tuner range). Conversely, a power stage simple enough to
realise the aforementioned low cost will have long switching
times (100ns) and use an unregulated power supply. The resulting
distortion will be well above 1% (even at low output powers) and
the output signal will be modulated by the 100Hz ripple on the
power supply. This will be very audible as power-modulated hum
in the output signal.

Unless
such low performance is acceptable, a simple analogue class D with
analog control plus a very cheap DA convertor (often offered
on-board DSPs as well) will provide much higher performance at the
same cost. This means that the argument of "digital all
the way improves signal integrity because of elimination of DA
conversion" does not hold true.
The reason is that the
power stage has extremely analog behavior, and it is easier and more
economical to control nonideal behavior in the small-signal path
instead of the large signal path, in the same way as loudspeaker crossover
filtering is easier to do in the small-signal path (active).

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Are class D
amplifiers as reliable as conventional amplifiers?

There is no reason why class
D amplifiers should be any less reliable than linear amplifiers. Some
people have expressed wonder over whether they never mis-switch.
They don't. Unlike software, hardware does not have a propensity to
crash or hang up - unless when overheated... which takes us to the
main topic:
The 3 most important factors dictating reliability
in any electronic product are temperature, thermal
behaviour and heat management. I hope I have your
attention now. Class D amplifiers are used for their low heat output
but this is not an excuse for improper heatsinking. In their drive
to demonstrate the efficiency of their products many class D vendors
underdesign their heatsinks or dispense with any altogether. Yet, by
the laws of thermodynamics a very small quantity of heat in a very
small heatsink translates in very high temperatures. As a result
many class D executions are unable to sustain their rated output
power for much longer than necessary to quickly measure it. This is done
under the premise that real music never drives an amplifier into rated
output power for longer than a few seconds. Unfortunately the
ingenuity of the proverbial User in abusing their equipment is
unlimited and many properly engineered products are known to have
burned out in their hands.

For this and other reasons
many countries (including the US) have regulation in place requiring
audio products to be able to deliver their rated output power for at
least 5 minutes. In terms of the physically small class D amplifiers
this is equivalent to "indefinitely". Don't go for less.

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Why is a ClassD
amplifier a potential EMI source ? What can be done about it? Is it
difficult?

EMI is caused by rapid
changes in voltages and currents. When the power stage is made to
transition (a process taking anywhere between 1ns and 200ns) the switch
outputs change across the entire power supply voltage and (barring one
exception) the entire loudspeaker current is re-routed through the
output stage. This is the root cause of EMI in class D amplifiers
and it is inevitable as well. Contrary to popular belief, the
voltage change (dV/dt) is not a big issue as long as capacitively
coupled currents can be returned directly to the source using
electrostatic shielding. High dI/dt values on the other hand will
readily provoke magnetic radiation, which in the far field turns
into electromagnetic waves. Working this EMI mode is done on 3
fronts:

Minimise dI/dt as much as
possible, i.e. relax switching speeds (if distortion is
controlled by feedback). Luckily the belief that high efficiency
requires blazingly fast switching is quite incorrect. A deep
understanding in MOSFET physics helps to determine correct
operating parameters.

Puzzle your lay-out
meticulously. It is not uncommon and totally acceptable to work on a
new power stage lay-out (only the switches, drivers and output
filter) for several weeks on end. After that, exercise similar
care about the rest of the board lay-out.

Be very careful about
using standard "EMI remedies". Most work counterintuitively and hence
tend to be counteractive (or have no effect at best).

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What is the
difference between meeting legal EMC requirements and being tuner
compatible?

Legal EMI requirements are
quite relaxed. The main idea is thou shalt not disturb thy
neighbour, where the latter is presumed to be well outside breathing
distance. To this end all equipment is required to sustain full
specified operation at a certain irradiation level and that it may
not produce radiation which at 3 metres distance would exceed the
same level minus a 10-20dB safety margin. Legal EMI tests should be
performed with the amplifier driven to 1/8 of its rated output
power.

"Tuner compatibility" is a
loosely used term to say that a [power amplifier] module inside a
product does not disturb the operation of the AM/FM/TV tuner inside
that same product.
The most important interference mode is noise
radiated outward and back into the external antenna. This mode can
be tested with a bare amplifier module and an external tuner, the
requirement being that the effective sensitivity (antenna level
required to obtain a specified SNR at the tuner output) is not
significantly affected. Such problems can be solved only on the
amplifier circuit itself.
Internal interference modes are
conducted (via ground and power supplies) and radiated from board to
board. Experience has shown that these can be worked either on the
amplifier, the tuner or on the product architecture. A designer
well-versed in EMI theory who has full control over the entire
architecture and layout can make a tuner operate fine together with a
perfectly foul amplifier. For real-world product designers this is not an
option so in this context we should define "tuner compatibility" as
being integratable by a normal product designer into a fairly
standard architecture.
AM-interference can be avoided to a large
extent when the switching frequency can be controlled and is kept
away from the receiving frequency. In TV and FM bands only good EMI
control will do and about 20dB extra margin should be calculated
relative to the legal limits. The output power up to which this
should be true is determined by the applicant. The same 1/8 of rated
output power should be a healthy guess.

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What are the
pro's and con's of  having class-D amplifiers with digital
input interface ?

In the case of real digital
amplifiers the interface will be digital by default. This question
concerns analogue class D amplifier subcircuits or modules with local
D/A converter. This option would be considered mainly for cases
where disturbances on analogue input signals are expected. Moving
the D/A converter away from the DSP section and onto the amplifier
indeed excludes any interference on the analogue signals. Care
should be taken before deciding for digital input though.
Transmitting the high-speed data and clock from the digital circuit
to the amplifier section is likely to produce a fair amount of
radiation. Coupling of data and other noise into the clock will
cause jitter, making the option unsuitable for real high-quality
audio.
Reducing the radiation would either require slowing down
transitions (further degrading jitter performance) or moving to
differential low-voltage signalling. The latter would probably both
reduce radiation and maintain good jitter performance but is quite
complicated and expensive. A more effective method seems to be to transmit
the analogue signal together with its reference (D/A converter local
ground) and use a simple op-amp+4 resistor circuit ("diff amp" or
"translator") to translate the signal to the amplifier's local
reference.

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Scanning the
market for various high-efficiency amplifier solutions yields a host of
names. What are the technologies behind them and what do they
offer?

Sharp
1-bit

Over the past few years Sharp
has been working the Japanese audio market with a type of class D
amplifier called "1-bit amplifier", riding on the waves of the 1-bit
audio (DSD-SACD) hype. Apparently they have gathered large following
in the industry.
The 1-bit amplifier operates like a 1-bit A/D
converter where the analogue input signal is summed with the output
from the switching power stage to form an error value. The error
value enters a high-order loop filter (in Sharp's case a 7th order)
to form a correction value. The correction value is quantised to 1
bit and sampled at a clock rate of 2.8224MHz, effectively yielding a
1-bit data stream at this bit rate. This bit stream controls the
power stage. The principal advantages of this technique are:

Very high distortion
rejection of >120dB across the audio band owing to the high-order
loop filter.

No loop gain compensation
is required with power supply variations (a one-bit quantiser has
undefined gain)

The
principal disadvantages are:

The undistorted output
swing is hardly more than 50% of the power supply voltage, limiting
available power to just over 25% of the normal maximum (with
switching losses virtually the same as for maximum power).
Larger signal levels cause progressive overload on the loop
filter resulting in noise-like distortion.

The high switching rate
causes increased switching losses.

These
two problems add up to an efficiency which is only as good as more
efficient class B power amplifiers.

The vendor also quotes the
"spread spectrum" nature of the switching waveform and hence the EMI
pattern. This is incorrect: 1-bit convertors (deltasigma converters)
have a strong discrete frequcncy component idling at half the
sampling rate and varying linearly (!) downward with absolute
modulation index. The inordinate amount of shielding on the vendor's
commercial products bears witness of remaining EMI problems.

This amplifier is touted as a
digital amplifier for direct DSD amplification. It isn't, for the
following reasons:

The input signal is
analogue. Even when a DSD source is used, it is effectively converted
to analogue using analogue lowpass filtering prior to entering
the amplifier.

The control loop is an
analogue feedback loop.

The "bit-stream" is not an
accurate representation of the wanted signal. Instead, by the
action of the feedback loop it contains the wanted signal plus
"compensation distortion" which will cancel the distortion
previously created in the power stage.

The power stage is
operating at a high switching frequency, exacerbating the relative
impact of the timing errors.

The power supply is not
regulated.

A remaining note concerns the
distortion. It would be reasonable to expect better than -120dB
distortion figures owing to the control loop's gain (120dB). In
reality, figures along the lines of -70...-80dB are found. One
should wonder what the open-loop distortion of the system would be... +40
dB (10000%)?
(To save the reader from sleepless nights: this
distortion is caused outside the control loop by the toroidal inductors
in the output filter)

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All-Digital:
TACT, TI, Pulsus and others

I have grouped these names
together (and might have forgotten a handful) as they share a common
take: open-loop amplification and fully digital PWM generation. As
said earlier, TACT on their Millennium product have pulled it off
very successfully indeed, with distortion figures consistently below
0.02%. Not surprisingly, this is done by thorough work on every
detail. The power supply is regulated and has an output impedance of
roughly 4milliOhms. The dead time is less than 10ns. The latter goes
at the expense of idle losses (and hence somewhat of efficiency) and
definitely EMI but what the heck, they pulled the trick. This
product also shows that getting good performance out of this mode of
operation comes at a cost: complicated drive circuitry, very tough
lay-out work, nonsaturating (i.e. huge) filter coils and a regulated
power supply. All in all a lot of analogue design expertise. A real
digital amplifier is not for everyone.
TI has purchased Tact in
order to make this technology available as chips. Doing so with the
PWM generator is a cinch, given Tact's fine VHDL work. Making power
stage chips to deliver the kind of performance to make "digital"
real is quite another matter. Worse still, the power supply will most
certainly be of a garden variety and will spoil what's left of decent
audio performance. Otherwise put, a digital PWM generator does not make
for a good digital amplifier, if at all worth the name...
The
digital PWM generator is the easy part of a digital amplifier,
which goes a long way explaing why so many people are into
it.

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Apogee
DDX

Apogee DDX is another player
in the All-digital league but it commands a separate entry. Not for
their intelligence alas... The DDX full-bridge power stage is
operated in a kind of 3-level mode (class BD), but not in bi-phase
PWM. The DDX power stage does offer the same efficiency advantage as
other class BD implementations but it introduces a new, previously
wholly nonexistent problem.
Around zero modulation bi-phase PWM
produces two nearly 50% switching waveforms which are practically
in-phase. The difference signal (as seen by the load) looks like
extremely narrow spikes, even though they are not physically created
as such (rising and falling edges of the narrow pulses are on
alternate half bridges). Apogee DDX explicitly try to produce these
very narrow impulses physically (having rising and falling edges performed
by the same half bridge with the other half bridge static). This
obviously a tough job, akin to modulating a normal PWM system near
full scale. The digital noise shaper is therefore amended with an
approximate model of the timing and amplitude errors committed
trying to get those narrow spikes out. The resulting system is
enormously complicated and does not perform any better than a
straight digital amp without these problems.

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Crystal
Semiconductors (Cirrus Logic)

At this stage they're "just"
another runner in the full-digital race, with the first chips
offering hardly more than a digital PWM generator. Their act is more
interesting than the others though, owing to their teaming up with
International Rectifier to develop gate drivers and MOSFETs
specifically targeted at class D. According to representatives,
Crystal realise that commercial products will in fact require a more
tolerant amplifier concept than a plain open loop system. Instead of
moving to analogue (simpler and more effective but far less sexy)
they are working to implement correction measures for the two chief
distortion mechanisms. This consists of monitoring the power supply
voltage with an A/D converter and measuring the switching delay
error, correcting for both in the digital domain. With due care I
should expect such a design to perform in the -80dB THD region,
which is indeed what they are claiming. While this system is definitely
the closest thinkable cross between digital amplification and analogue
realpolitik, there is more analogue processing to it than to
a straight D/A converter plus analogue class D amplifier.

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Bang &
Olufsen PowerHouse, ICE Power

B&O shun the term class
D, instead calling their amplifiers Intelligent, Compact and
Efficient, conveniently acronymating as "Ice". Their most important
products are the 250 and 500W "sandwich" modules, based on a
properly optimised full-bridge power stage operating in a
self-oscillating mode called Phase Shift Controlled PWM (PSCPWM), in
which the idle frequency is that at which the combined phase shift
of output filter, loop filter + compensation and power stage delay
goes 360°. The loop filter is high-order and an on-board zobel
network keeps the system stable under odd or no-load conditions.
Feedback is taken after the output filter, resulting in low output
impedance and good rejection of distortion caused by the toroidal output
choke. Performance figures are very respectable (among the best on the
market), though independent assessment rates the sound performance
as less than high fidelity. EMI on the module is within legal limits
but only just, potentially resulting in problems on the product
level.
Other offerings and developments by this company are
kilowatt power amplifiers operating in bi-phase mode (dubbed class
BD) and pseudo-digital amplifiers (dubbed PEDEC). Class BD reduces
switching frequency and hence switching losses by a factor of 2 and
improves open-loop distortion by a factor of 2 as well. The PWM
drive is based on a fixed clock frequency. This allows for less
tight control over output impedance but otherwise performance
figures are still very respectable.
PEDEC is based on a digital PWM
generator and an analog feedback loop that corrects for timing errors
by adjusting the edges of the incoming PWM.

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Class T,
Tripath.

Tripath employs a rather
strict definition of class D (namely a fixed-frequency PWM based one) in
order to award themselves a new class name, class T. Nonetheless, for
all intents and purposes it may be classified as class D without
omission.
They were the first to deliver a serious chip to the
class D scene, a 10W device targeted at multimedia. Another product
is a driver module to build higher power amplifiers with. Power
stages found in Tripath's products are half- and full-bridge setups.
The control system is a 3rd order switched-cap loop controller (the
first order forcibly converted to continuous-time to prevent
aliasing of switching errors and noise) with hysteresis modulation
to create a time-quantised PWM (of variable repetition rate) signal.
The time-quantisation implies that the PWM will have quantisation
noise on it. For a third order loop (now a noise-shaper) to
sufficiently control this noise inside the audio band a high sampling
rate (switching frequency) is required. Indeed, the (idle) switching rate
is frighteningly high at 1.4MHz. The result is that even Tripath are
reluctant to claim much better than 80% efficiency.

Distortion figures are good,
but are usually spoiled by the coils of the output filter which is
outside the control loop. The figures inspire Tripath to make claims
toward "audiophile" sound quality, but support of this notion is not
unequivocal.

EMI of Tripath products is
reportedly troublesome, with the company unwilling/unable to offer
application support for EMI problems.

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Jam, Class
J

Technology start-up Jam has
come up with a more unusual power stage approach, namely one which can
produce voltages both below and above the supply voltage (i.e. a
buck-boost converter). Although class D by the definition given earlier
it is very different from the text-book power stage, meriting the
company their new class name more than Tripath theirs. The basic big
thing about it is that it'll put out higher voltages than the power
supply rail, allowing high output powers into normal load impedance
out of low supply voltages.
Unfortunately, while innovative the
technology does not offer a practical breakthrough. Some
back-of-envelope work suggests that the number of switching
components remains identical to that of a boost converter plus a
normal class D, with silicon area also identical.
The company
prefers to operate the power stage with an open-loop digital
controller, but this is not a defining feature of Class J.

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Philips

Philips is a fragmented
company and not surprisingly, has several class D products developed
quasi-independently.

TDA89xx

These are two-chip amplifiers
(one stereo modulator and one stereo 100W power stage) based on more
or less text-book PWM. The loop filter is second order and together
with the tightly switching power stage, offers good THD results
(rising at higher frequencies). The output filter is outside the
control loop. EMI results are among the best on the chip market,
with Philips Semiconductor's application boards operating very
comfortably below any conceivable legal limits (barring automotive
maybe).
Power ratings are quite optimistically specified,
requiring a bone-hard power supply delivering the chip's absolute
maximum rating.

SODA

This is a discrete circuit
with a half-bridge power stage operating in a hysteresis switching
(self-oscillating) mode. Feedback is combined from before and after the
output filter. Interestingly the entire modulator/feedback chain is
built with only passive components directly connecting to the
comparator. MOSFET drivers are a discrete affair and the power stage
consists of one N-channel and one P-channel MOSFET. Distortion is
commendable and virtually independent of frequency, making THD at
10kHz among the lowest seen (0.02%). Output impedance is
significantly better than that of designs without feedback take-off
after the output filter (the majority) but higher than the B&O
modules.
In listening tests, the sound quality of the design
lab's own 2x100W sample was consistently found to be quite stunning,
sitting among devices whose cost per channel alone warrants a
mortgage.
EMI is comfortably below the legal limit.
The
circuit seems quite fiddly in optimisation so it would be best to copy it
straight from the reference design and resist making changes to it
for whatever reason. The technology can be licensed from Philips but
given the above it would seem wisest to try and persuade them to
make modules for you (the company not having announced any formal
OEM module plans yet)

UCD

This circuit is intended to
replace SODA in the short run. Somewhat immodestly called "Ultimate
Class D", it sports milliohm scale output impedance (even at 20kHz),
very good THD figures and a healthy efficiency that could allow it
to operate without a heatsink in normal operation. Yet it has one.
The circuit is a half bridge with two N-MOS this time and fully
discrete drive circuitry (including the comparator).
The
modulation scheme is a Phase-Controlled self-oscillating design with
feedback taken only after the output filter. Around this, one or two
further orders of control are built. EMI results have yet to emerge,
but supposedly the target is to improve on their previous work. As
this is a discrete circuit even more than SODA, buying modules is
advisable.

PP-A (Phased Power
Analogue)

PP-A is a no-holds-barred
design that uses a 4-phase (5-level) power stage driven by 6 active and
2 passive orders of loop control (the 2 passive ones apparently
accounting for the output filter). Combination of these 4 switching
signals is done using a patented inductor-summation "tree" built
around center-tapped coils as nodes. While the control of the 4
power stages requires a large amount of digital logic, the amplifier
is definitely analogue. To contrast this design with Sharp's, it
uses the multilevel output stage to improve undistorted modulation
index to 88% and to reduce switching frequency to a manageable level
(around 200kHz, promising very good efficiency). Also, the control
loop operates solely on the basis of the filter output, promising
negligible output impedance and distortion. Nicer still, the
spectral behaviour of multilevel noise shapers is truly spread-spectrum
with dominant tones virtually absent.
The concept suggests
ridiculously low distortion figures could be attained (-130dB??),
but one should remain skeptical of the importance of such low THD.

A discrete implementation is not for the financially faint of
heart (i.e. only for the audiophile community), mainly because of
the 4 output stages and the FPGA based logic block. A chip version
is planned that should significantly slash costs while retaining
most of the performance.

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Trying to
measure distortion figures gives me nowhere the kind of results that
vendors are specifying. Why is this?

You are probably measuring
distortion and noise with the switching residual still present. The
output filter of a class D amplifier is intended to reduce the HF
switching noise such that it no longer produces significant
dissipation in the intended load, not to reduce it until it vanishes
in the noise floor. When making measurements a measurement filter
should be inserted that does attenuate the switching component ad
oblivion. Normally the 22kHz or 30kHz filter fitted as standard on
most audio analysers should do the job. A side-effect is that THD
readings suddenly "improve" beyond 7kHz as the dominant harmonic
(third) moves out of band. Extrapolating the readings is generally
valid.

A minor snag on the Audio
Precision analysers is that the autoranging circuitry on the "reading"
detector (which sits after the filter) is set on the basis of levels
measured before the filter, resulting in the distortion only occupying
lowest few bits of the post-detector-A/D. This gives typically
"jumpy" readings at low amplitudes (the instrument sometimes failing
to put out a reading altogether). The only option is to defeat the
autoranging feature.

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How can I infer
sound quality from measurement data?

The best way to determine the
sound quality of an amplifier is to hook up a source and a pair of
good speakers and listen. Some even go so far as to claim
measurements are uncorrelated with audible quality. The reality is
"yes and no".
Yes, there are amplifiers with excellent
figures that put out a perfectly foul sound. And yes, there
are amplifiers with very mediocre figures that sound quite fabulous.
But no, there are some important things to be learned from
measurements.

Firstly, it is true that
absolute distortion figures tell little about how an amp will sound. The
behaviour of distortion versus frequency does.
Assume a power
stage operated in "open loop" (this may be hypothetical, not all
amplifiers can work in open loop at all). For such a power stage,
the ideal condition would be the one which has the same voltage
transfer all the time, regardless of the rate of change of the
signal processed. Such an amplifier will have THD totally
independent of frequency. Typical examples are zero-feedback triode
amplifiers but also some very carefully designed zero [overall]
feedback solid-state amplifiers. Both categories are known to offer
fabulous sound, with higher distortion levels only producing a
certain "colouration" without affecting transparency.

An open-loop amplifier that
produces frequency-dependent (ie. increasing with frequency)
distortion levels has underdesigned driver stages (in linear
amplifiers) or badly chosen output coils (in class D amplifiers,
such as toroids or too small ferrites). Distortion typically rises
at a rate of 6dB/oct. Amplifiers with such problems can still attain
good (but not excellent) sound performance, but with varying success
and dependent of the type of music used.

When feedback (first order
assumed) is next applied, distortion products will be attenuated better
at frequencies where loop gain is high (low frequencies normally),
and will remain progressively higher at increasing frequencies.

Amplifiers that had a flat distortion characteristic in the open
loop will then exhibit a distortion characteristic rising at
6dB/oct. Even though the distortion will have diminished, the
amplifier will have acquired the glassy, sticky sound, which seems
only to worsen as the dominant pole nears DC, and which is commonly
known as "the negative feedback sound". Practice has shown that it
is wiser to limit loop gain such that it does not rise any further
below 20kHz. This indeed retains the "zero-feedback sound" while
simultaneously improving significantly on neutrality.
Of course, this
is bad news for specmanship. Usually distortion at 1kHz is the most
proudly noted number on an amplifier's data sheet, whereas the 10kHz
distortion (typically 20dB worse) is tucked away in a graph that may
or may not have been reproduced.
Now, describing the sound of an
amplifier with rising distortion in the open loop with extra
feedback added is left as an exercise to the reader.

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How do class D
amplifiers sound?

OK, I admit. I edited that
question. Class D amplifiers have a reputation for unsatisfactory sound
quality. Presumably this is caused by a couple of very early
products that were released in the 70's and 80's and that sounded
quite abysmal. The current state of technology is quite different.
Practically any half-decent implementation of a simple class D
amplifier executed using today's components sounds warm,
dynamic and musical, having the kind of direct appeal
also found in vacuum tube amplifiers. It is this discovery combined
with the previous bad reputation that has led most budding class D
designers to conclude they had something wonderful and "audiophile"
on their hands (and proclaim it in their flyers). Unfortunately, an
appealing sound is just a fraction of what constitutes truly
audiophile-quality sound (neutrality and transparency
to name a few). Of the class D amps auditioned so far only 2 pulled it off
(incidentally the ones with a flat THD response).

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Watt's behind
the power rating?

(Sorry, stupid pun) The power
rating of finished and boxed amplifiers is already a contentious
issue, let alone that of chips and modules. Let's have a look at the
product case first:
One amplifier could have the following power
ratings:

50W 1%, 60Hz-20kHz
(IHF)

65W 10% THD (EIAJ)

80W Music Power
(DIN)

130W Peak Power
(Sales)

1300W PMPO (Yahu
Wahu)

where:

IHF, the Institute
of High Fidelity tests rated power by first warming up the
amplifier by letting it cook at 1/3 of the rated power for an
hour and then testing, at rated power, for 5 minutes, if the
amplifier is capable of delivering quoted distortion over the
quoted test band (whereby strictly speaking THD is measured by
harmonics only). This is the mandatory rating system for
products sold in the US (enforced by the FTC).

EIAJ allows power
to be rated on a non-warmed-up device, right after power-on. The
output power at 1kHz for 10% THD (heavily clipping), is the
rated power.

Music power tries
to fathom dynamic behaviour of the power supply by measuring
average ("RMS") power on a single-cycle 1ms tone burst.

Peak power is twice
the average power, assuming a non-clipped sinewave. Even if
average power was measured at 10% THD this assumption is
made.

PMPO ("Product
Manager's Power Output") is a nondescript item that allows any number
to be attached to it.

Things are not that sticky
when dealing with chip or module vendors. Still there are a few
caveats:

What power supply voltage
is stated for rated output? All too often it is the breakdown
voltage of the device. Real life power supplies (especially
transformer-based ones) have outputs varying according to load
and line conditions. Check very carefully what your power supply
will deliver under nominal-line, full-load conditions given that
it should never exceed the breakdown voltage at line overvoltage
and no-load conditions. Derate the power according to the square
of the ratio between those two. With chips and a traditional
power supply you'll quite often end up cutting power rating by
half. Switchmode power supplies normally have much better regulation,
allowing higher output power to be delivered reliably.

What distortion spec is
used to rate power at?

Can the amplifier deliver
rated power long enough to pass the tests in vigour in the areas
where you will sell your product?

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I asked my
vendor if he couldn't re-engineer his 250W/4 ohms power amp for
250W/8 ohms and he swallowed uneasily. What's the
problem?

Well, 250W into 4 ohms
requires a peak voltage of roughly 44V. With some margin this can be done
with 55V or 60V MOSFETs. To get 250W into 8 ohms requires at least
63V, which means 100V MOSFETs. The MOSFET market is not driven by
class D amplifiers, but by computers and cars. This means that the
low-voltage fets are usually far more evolved technically than the
high voltage parts. It may well be that the move from a 60V to a
100V FET pushes your vendor's existing design over the EMI limits,
and increases switching losses by a factor of two. In general, if
someone has a properly functioning unit with certain power and load
specs it is often more economic to modify the application than it is
to modify the amplifier module.
This is quite a difference from
linear amplifiers where scaling is just a matter of changing output
devices or adding some more.

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Where does one
start to select an amplifier or vendor for a specific
application?

Get down to the tough matters
first. In which aspect are you requiring extreme performance from
your amplifier (or your vendor)?

EMI

Sound quality

Efficiency

Power

Tech appeal

Cost

Time to market

(etc).

Start with the one you
value most. Before you're 3 items down your list you will have
pinpointed your vendor (and possibly the product too).

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