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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