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Titolo: SACD history: Super Audio-CD "The Rich Report"

da: 
http://www.stereophile.com/features/374/index.html
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By David Rich   �   November, 2000
Although Philips invented the Compact Disc, it was only when Sony got
involved in the early 1980s that it was decided�at the prompting of
conductor Herbert von Karajan, a close friend of Sony's then-president
Akio Morita�that the CD should have a long enough playing time to fit
Beethoven's Ninth Symphony on a single disc (footnote 1). Even if the
conductor was using very slow tempos, and even given the minimum pit
size and track pitch printable at the time, the 16-bit data and
44.1kHz sampling rate they settled on gave them a little margin.
From the beginning, it was pointed out that Linear Pulse-Code
Modulation (LPCM) encoding at 16-bit/44.1kHz may not have been good
enough to be audibly transparent. Research indicated that a
signal/noise ratio of greater than 98dB would be needed before the ear
would discern any artifacts in its most sensitive range. In addition,
the choice of a 44.1kHz sample rate left a transition or guard band of
just 2kHz between the audio passband and the stopband for the
anti-alias (record) and reconstruction (playback) filters.
This narrowness of the guard band has always been controversial. With
a 44.1kHz sampling frequency, signals higher in frequency than
22.05kHz will alias�ie, produce enharmonic image products in the
audioband�and thus need to be filtered out. This requires very steep
low-pass filters that can, some papers claim, cause audible problems.
The 48kHz sampling rate of DAT recorders and DVD-Audio players doubles
the guard band, but it is not widely reported how this small increase
in sampling rate could make such a big difference in perceived sound
quality. In recent AES preprints, the most optimistic assumptions
about human hearing put the maximum sample-rate requirement at about
60kHz (footnote 2).
Now the questions about digital technology as embodied in the CD
format are about to disappear. The 4.7-gigabyte capacity of the
DVD-Audio disc, achieved by a smaller pit size, closer track pitch,
and a laser of slightly shorter wavelength, can give us 5.1 channels
with a 24-bit depth, and 24 bits give a maximum S/N ratio of 144dB.
Distortion is at that level too. Such low levels of noise and
distortion are well beyond the limits of human hearing�and even beyond
the limits of electronics, for such fundamental reasons as thermal
noise in circuits and transducers.
On the sampling-rate front, 96kHz gives us a guard band of 28kHz
between the nominal 20kHz top of the audioband and the Nyquist
Frequency of 48kHz. This can be used to make the anti-alias filter
less sharp�ie, with a less steep initial rolloff�which will reduce
pre- and post-ringing on impulses. Alternatively, by staying with a
2kHz guard band, we can record up to 46kHz at 144dB signal/noise, if
you think you can hear that high (footnote 3). (Papers have shown that
some young people can hear past 20kHz.)
Any way you slice it, its 24-bit data size and the 96kHz sampling rate
possible for 5.1-channel recordings puts DVD-A well beyond the limits
of audible problems. However, the DVD-A launch was delayed until
copy-protection issues had been worked out. [And even with players
available as of summer 2000, the copy-protection issues had not been
fully resolved.�Ed.] The good news is that, because it is based on the
Digital Versatile Disc technology, which benefits from significant
economies of scale, the first-generation DVD-A players will be
relatively inexpensive.
Meanwhile, there is another competitive standard to consider: the
Sony/Philips Super Audio CD. Rather than use LPCM, SACD uses a
proprietary fixed delta-sigma modulation code, known as DSD. As
delta-sigma modulation technology is well-known to audiophiles, I will
only briefly summarize it here.
In the simplest form of delta-sigma modulation, a 1-bit converter is
oversampled in a feedback loop. By definition, a 1-bit system�it's
either on or off�suffers from very high quantization noise. However,
the feedback loop shapes the noise energy so that noise is acceptably
low in the audible band but high outside it.
Audiophiles have been suspicious of negative feedback for years, but
for reasons I don't understand, they appear to have given delta-sigma
modulation their implicit approval. Unlike feedback loops in
amplifiers, where the loop circuit is already somewhat linear, the
feedback loop in a delta-sigma converter is wrapped around the most
nonlinear component we can imagine: a 1-bit A/D converter.
Why would anybody do this? As two points define a straight line, by
definition a 1-bit converter has zero linearity error. With
noise-shaping we can have a converter with as many bits' worth of
effective resolution as we like, as long as our oversampling rate is
high enough and the noise-shaper is of a high enough order. The
delta-sigma modulator is an effective way to make high-resolution
converters cheap by replacing complex, multibit ADCs and DACs with
simple, cheap, single-bit units. In the real world, of course, that
simple converter is very complex to design: noise and distortion must
be considered in the ADC's noise-shaper and sampler, and in the
charge-to-voltage converter and reconstruction filter of the DAC.
We get no free lunch with a delta-sigma modulator. Using a high degree
of negative feedback around a strong nonlinearity can give rise to
idle-channel tones and limit cycles. (In effect, the converter behaves
as an oscillator�it puts out a repeating pattern even when there is no
input.) Numerous papers have been published in both the AES and IEEE
Journals describing this phenomenon. In particular, the difficulty in
analyzing, modeling, and eliminating these non-ideal effects has been
described.
These quasi-periodic oscillations have been shown to be audible, at
least with test signals, by me and Steven Norsworthy in the AES
preprint mentioned in the "References" sidebar. They are easy to hear
even using Matlab computer simulations (output files can be played
through the computer's soundcard), but their quasi-periodic nature
makes them hard to measure, even in computer simulations in the
frequency domain. In that paper, and in previous work by Steve, it was
shown that the tones can be reduced by increasing the oversampling
rate and by using a coder of more than 1-bit quantization. In fact,
all high-performance ADCs using delta-sigma modulation now use a
quantizer with at least three codes (Crystal), four codes (AKM), or
five codes (dCS). Oversampling rates are universally 128x (relative to
44.1kHz) or more.
It should also be noted that 1-bit delta-sigma bitstreams are very
sensitive to such analog issues as clock jitter, edge speeds, and
folding of the very-high-energy out-of-band noise into the baseband.
This gives rise to more tones and oscillations. The multi-level
approach used in all high-performance delta-sigma DACs these days does
reduce this problem, however.
In delta-sigma DAC designs the trend has been to go to 20 or more
levels for the internal DAC instead of one level, and up to 18-bit
linearity (Analog Devices, Crystal, and even single Bitstream inventor
Philips Semiconductor). An advanced technique that randomizes a
multibit DAC's linearity errors by shuffling the DAC elements around
for each code conversion has been universally employed for high-end
converters. Oversampling rates of 128x are now universally used for
the highest-performance DACs, as they were in the ADCs. Burr-Brown
continues to use full 20-bit sign-magnitude DACs for its
highest-performance products, but these products use no feedback,
instead relying on analog matching techniques to achieve the desired
linearity.
In terms of information theory, a delta-sigma stream can be shown to
carry less information than a linear PCM stream of the same
bandwidth/noise-floor product (footnote 4). It makes sense, therefore,
to convert the bitstream output by a delta-sigma ADC to an LPCM stream
before it is sent to a storage medium. In digital signal-processing,
the circuit block that converts a high-speed low-bit
datastream�output, for example, by an ADC�to a low-speed, higher-bit
stream, is called a decimator. The decimator reduces the total number
of data bits per second (bit depth times the sampling rate) coming
from the ADC. Thus the decimator changes the inefficient delta-sigma
stream from the ADC to the efficient PCM stream. (In the case of 24/96
DVD-A, this is 2.3 megabits per second per channel (footnote 5).
The equivalent DSP block in a DAC design is called an interpolator; it
converts the incoming low-speed multibit stream to a higher data rate
to drive a delta-sigma DAC. In effect, the efficient PCM stream is
turned into the inefficient delta-sigma stream, which runs at a higher
rate of total bits per second. Remember, the reason we create a
delta-sigma stream is because it is cheap and easy to make DACs and
ADCs that have low resolution but that run very fast with inherently
good linearity. It is difficult and expensive to make lower-speed,
higher-resolution converters that work directly on the PCM stream.
This is the only reason for the existence of the inefficient
delta-sigma stream.
With 0.12�m CMOS integrated-circuit technology, well-designed
interpolators and decimators are so small they cost almost nothing.
However, note that I said "well-designed." All the digital arithmetic
must be properly done, including data-rounding and addition of dither.
Internal data paths and filter coefficients must be as wide as
necessary to ensure that no nonlinearities are introduced.
Please note that it is possible to build a decimator or interpolator
that can distort the signal. Properly designed decimators and
interpolators are audibly transparent, but improperly designed ones
can introduce audible errors. I have little doubt that low-end digital
filters/interpolators in some CD players are cheapened enough to cause
audible problems. But bad design must not be confused with a
fundamental problem in the concepts of decimation and interpolation.
With Super Audio CD, Sony and Philips offer a product that encodes the
inefficient high-sample-rate, low-resolution bitstream. The Direct
Stream Digital (DSD) encoding used on SACD is nothing more than the
raw bitstream output by a seventh-order delta-sigma A/D converter. DSD
uses only a single bit to encode the signal, whereas the
high-performance ADCs from all leading manufacturers�and all work in
advanced research�use three codes, and in many cases more than that.
Significantly, the Sony/Philips DSD system oversamples only at a 64x
rate (relative to 44.1kHz). That's half the rate used in all
high-performance ADCs and DACs available today. To get better than
20-bit performance below 20kHz with a 1-bit stream, a seventh-order
modulator must be used at this low oversampling rate. The higher the
modulator order and the lower the number of levels, the more difficult
it is to remove the high-frequency energy of the 1-bit quantizer. The
problems made worse include sensitivity to clock jitter and matching
of clock edge rise and fall times. Aliasing of out-of-band noise into
the passband becomes more difficult to deal with in a high-order,
single-bit system.
The one-bit encoding and 64x oversampling rate results in a data rate
of 2.8224MHz per channel for DSD, which looks similar to 24/96 LPCM's
rate of 2.3Mb/s. Remember, however, that DSD uses an inefficient
delta-sigma bitstream. If we look at the theoretical spectrum of a
full-scale, DSD-encoded 1kHz tone (fig.1, after Sony), we see that the
24-bit performance is not achieved in-band, and that the S/N ratio
drops quickly above 20kHz as the quantization energy from the 1-bit
coder becomes significant. The 24/96 LPCM used on DVD-A offers 24-bit
noise and distortion performance all the way up to 46kHz.
Fig.1 DSD encoding, spectral analysis, DC-200kHz, 1kHz tone at 0dBFS
(log. frequency scale, FFT bin width 10Hz). After Sony.
Because DSD uses a high-order delta-sigma modulator, the noise above
20kHz rises very quickly. (The higher the order, the faster the rise.)
To prevent high levels of high-frequency energy from getting out of
the player, an analog low-pass filter at 100kHz is required during
SACD mastering. Even so, a large amount of out-of-band noise might be
passed on to the power amplifier, perhaps as high as a tenth of the
full power output of the system. If not rolled off by the amplifier,
this may be just below the energy level to harm a tweeter. This is why
SACD players are required to use a further 50kHz low-pass filter on
their outputs (though this can be defeated). The need for these
low-pass filters works against claims that DSD has wide bandwidth and
low phase shift. 
Making the inherent data inefficiency of a delta-sigma datastream
worse is the fact that lossless coding is less effective than the MLP
used in the PCM-based DVD-A. Because of all the high-frequency energy
in the stream (energy that would be rejected by the decimators used in
a PCM-based system), the DSD data are less correlated and less
susceptible to efficient packing.
Why would Sony and Philips introduce a medium that, while offering
higher resolution than CD, uses an inefficient coding scheme, needs
great care to avoid stability problems with that coding, and requires
analog low-pass filtering to ensure there are no compatibility
problems with the playback system? Because, they claim, the DACs to
decode the stream can be made cheaply, and it might be possible to
pass the datastream all the way to the power amplifier.
Nice marketing talk, but this can never be allowed to happen: If the
1-bit stream came out of the box, it could be perfectly copied. It is
important to understand that the data stored on a SACD are encrypted,
and that the encryption must be reversed before the 1-bit stream can
be sent to the D/A converter. Sony's first SACD players, the SCD-1 and
SCD-777ES, have separate chips for their digital operations and their
DAC stages, and their product literature discusses the advantages,
beyond simple cost considerations, of putting these functions on
separate chips.
However, we can guess that the goal for SACD players is for the LSI
chip that removes the encryption to also produce the analog output.
That way, the in-the-clear datastream need never be taken outside that
chip. However, this is hardly a low-cost solution, as analog and
digital electronics must be made to coexist on the same chip, adding
manufacturing complexity. The problems Sony now overcomes through the
use of separate chips will have to be overcome in a different way, or
the degradation in performance may have to be lived with.
Note that Sony understands these problems clearly, and discusses them
in depth in its current literature on the SCD-1 and SCD-777ES. They
point out problems with a 1-bit system in what they call the amplitude
and time axes, and discuss complex and expensive solutions to overcome
the problem.
One very significant problem they do not discuss is the aliasing of
the very-high-level noise near half the sampling rate back to the
audio baseband. To achieve close to the theoretical performance of the
seventh-order modulator, interfering clock signals on the reference
voltages must be attenuated by as much as a million-fold if true
20-bit performance is to be achieved. This is clearly impossible. To
prevent noise or tones in the high-frequency region of the 1-bit
spectrum from being folded into the audio baseband, op-amp performance
must be exceptional. Nonlinearities in the op-amps, including those
caused by slewing, can cause this problem.
So if circuit complexity is not necessarily reduced by storing
DSD-encoded data on a SACD, with no cost savings to be found,
again�why would Sony and Philips do this? Here is my guess, informed
by some deep-background info I was given by a Sony employee:
Why use a 1-bit system rather than LPCM? SACD advocates like to point
out that the digital signal path is significantly simplified by the
elimination of the decimator, interpolator, and DAC noise-shaper.
However, this may be a cover to hide the real reason for moving to
SACD: storing the data as a single-bit datastream allows better
copy-protection than does PCM. Because all bits are equally important
in DSD�unlike in PCM, where the MSB is most important and the LSB is
least important�intentional bit errors, properly executed, would be
inaudible but detectable. Maybe they can be made detectable even after
being recorded on analog systems. These bit errors could be SACD's
equivalent of the perceptually coded "watermark" so often talked about
in connection with DVD-A, or it could be something else. (Information
is impossible to come by.) 
Why am I skeptical about the "simple signal path" explanation? In
addition to the practical problems of converting an SACD datastream
directly to analog that I discussed above, the proposed simplified
signal path assumes no signal processing between the digital tape
recorder and the SACD mastering process. Even something like mixing a
multitrack recording�mixing can occur directly on a delta-sigma
stream�results in a multibit stream, even if the mix occurs
one-to-one. While it is possible to equalize and even add reverb in
the delta-sigma domain, these again result in multibit streams. Once
we have a multibit stream, we must convert it back into a 1-bit stream
using a digital process similar to noise-shaping. These operations
will have to be repeated many times as cascades of EQ and mixing
operations occur.
Yes, pro audio companies are introducing the EQ and mixing tools to
allow engineers to work directly on material recorded in DSD. But it
is fair to point out that, unless studios are completely redesigned,
it is highly likely that SACD signals will be converted to PCM signals
to allow mixing, equalization, editing, etc., using current digital
equipment. And at the consumer's end, it is very difficult to
implement digital room-correction systems and digital loudspeaker
crossovers in the SACD domain. Even if this becomes possible, we will
once again be faced with a multibit stream that must be processed back
to a 1-bit stream.
Again, any processing of the 1-bit stream on the user's end assumes
that it will be available at the output of the SACD player. For
copyright reasons, this is something that might never be made
available.
What is clear is that the Sony/Philips Super Audio CD system does not
perform to the same level as the best LPCM converters now available.
The best ADCs' delta-sigma modulators, for example, produce four times
the data per second as DSD, which means a lower noise floor and lower
levels of distortion than can be produced with DSD (assuming the same
loop-filter order).
In addition, even though current chips can perform, at best, only at a
true 19-bit level, DVD-A offers a capacity of 24 bits. We can expect
the number of bits produced by the finest ADCs, and the sampling rates
at which they operate, to continue to increase in the future as chip
designers attempt to come as close as possible to the 24-bit
performance level. Advances in silicon-chip manufacturing that allow
the development of very fast microprocessors�including copper
interconnects, trench isolation, and high-energy implantation�can be
applied to speed the sampling rate of ADCs as well as allow them to
encode more bits per sample. In addition, the introduction of
very-high-speed BiCMOS-process technologies developed for
radio-frequency systems, including new silicon germanium bipolar
devices, will allow the analog circuits in delta-sigma-based ADCs to
work well at these higher sampling rates.
And keep this in mind: Burr-Brown already produces DACs that do 96kHz
sampling with 24-bit resolution�resolution, not the levels of noise or
distortion�and which use no feedback. Using the very-high-speed
converters designed for cellular radio applications, it is already
possible to do A/D conversion at the 18-bit level without significant
feedback. Would such converters give better sound than the delta-sigma
ADCs used to make current 24/96 recordings? (And these delta-sigma
ADCs, oversampling at twice DSD's rate and using at least three-code
quantizers within their delta-sigma loops, are already better than
DSD's 64Fs 1-bit converters.) We don't know the answers to these
questions.
While CD was introduced as "perfect sound forever," it had only two
channels of output, which put the lie to that statement right
away�clearly, multichannel systems were required to create the sound
of a good concert hall. Both SACD and DVD-A offer multiple channels;
but with DVD-A, the space available for experimentation and
improvements is almost infinite. With Super Audio CD, we are stuck
with a system that cannot be improved. It is likely that SACD was
brought to market not as a way of bringing audiophiles closer to the
music, but of making digital audio more difficult to copy. 
Footnote 1: I was present at the 1981 Salzburg, Austria press
conference where Morita-san first presented this explanation of why
Sony had asked for an increase in the planned disc size. (This was
also the occasion where HvK made his infamous "All else is gaslight"
quote about the CD.) Perhaps more important, Sony had persuaded their
Dutch partners that the planned 14-bit word length was insufficient to
achieve the desired sound quality. For me, the unsung hero of the CD
concept was Toshi Doi, who came up with the error-correction code and
was a forceful advocate for 16-bit word lengths. Dr. Doi is currently
a leading light in Sony's robotics research efforts.�John Atkinson
Footnote 2: See, for instance, Robert Stuart, "Coding Methods for
High-Resolution Recording Systems," presented at the 103rd Audio
Engineering Society Convention, New York, 1997. Available from the AES
as preprint #4639, and in expanded form from the Meridian website
library as Coding2.PDF .�John Atkinson
Footnote 3: For a discussion of this subject, see my "What's Going On
Up There?" in the October Stereophile, pp.63-73.�John Atkinson 
Footnote 4: See Malcolm Omar Hawksford, "Bitstream versus PCM debate
for high-density compact disc."�John Atkinson
Footnote 5: For five channels of 24/96 data, this results in a total
data of 11.5Mb/s, which is higher than the maximum possible rate that
can be retrieved from a DVD. The DVD-A specification allows this rate
to be reduced either by encoding the rear channels at a lower sample
rate, or by using Meridian Lossless Packing (MLP). MLP takes advantage
of the correlation between samples with music data to reduce the
amount of data encoded on the disc. Unlike lossy compression schemes,
such as Dolby Digital, or DTS at the lower rates, the PCM data packed
with MLP can be retrieved with no loss of information.�John Atkinson
**
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Aliis alia placent
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quote about the CD.)
Atkinson quando fu presentato il CD da Karajan e Morita (Sony) non ha
mai perdonato al mai troppo compianto direttore austriaco
quell'espressione (tradotto, sarebbe una specie di "tutto il resto �
fuffa") che contribu� a dare il colpo mortale alla concorrenza tutta.
E che fu un colpo mortale lo si not� col vertiginoso incremento di
vendite che il supporto argenteo digitale ebbe nei confronti del
vetusto padellone nero, che fu relegato negli archivi polverosi della
Storia.
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