THE COMPACT
DISC
Philips developed the basis for the current Compact Disc and Sony joined during
the last years when Matsushita decided not to work with the Philips engineers.
The Philips engineers developed the 120 mm (4.7 inches) disc which holds the original
signal and the various corrections which are necessary to read the stored signal
correctly. This certainly was no mean task. It is a clever storage medium.
Since Philips initially developed a converter with 14 bit quantization that worked
with the chosen sampling frequency of 44.1 kHz., Sony insisted that the quantization
should at least have 16 bit because severe criticism was expected, especially
from audiophiles and serious music lovers, who by the end of the nineteen seventies,
possessed a phenomenal sound reproduction of analogue recordings on records
and tapes:
-
a
high resolution of every frequency in the
audio band,
-
an
extended (practically unlimited) frequency band for upper harmonics, and
-
a
very good transient response,
-
good
dynamics,
-
an
extremely low noise level, and
-
minimal
distortion for the given format.
14 BIT and 16 BIT
Despite the advantages of digital recording, for
many the format of the Compact Disc of 44.1 kHz. and 16 bit was a serious disappointment
as far as resolution and frequency band are concerned. With 16 bits there are
65.536 pulses.
However Philips originally proposed a 14 bit quantization
resulting in 16.384 pulses per second. So 16 bit is quite an improvement.
NOISE
SHAPING
In order to be able to comply with the 16 bit criterion Philips
applied the technique of four times over sampling and incorporated a noise shaping
filter to its 14 bit converter. Noise shaping increases the dynamic range. The
addition of noise shaping resulted in an outcome which was closer to the 16 bit
dynamic range of 96 dB (each bit representing 6 dB).
The noise shaping however
is noticeable in these first Philips players and especially the high frequencies
do not have a clean defined signal but appear to sound somewhat diffuse, or, in
other words, show a noise which is not stemming from the audio signal itself.
Important in this matter however is the order of the noiseshaping filter with
it's audible effect. By reducing the level of noise shaping, the specific filter
can be less severe. But there always remains a difference between a data stream
with and one without noise shaping.
LOW
BIT
Another way of calculating the conversion of the signal is the so
called Low Bit Converter. The low bit converter does exist thanks to upsampling
(multiplying the data) and the (necessary) application of a noise shaping filter.
The advent of high speed processors made the creation of these so called low bit
or one-bit converters possible.
DISCMAN
Initially the low bit converters were meant for portable digital
equipment: CD-players and DAT recorders for outdoor use by joggers, skaters and
people traveling. Low bit converters are very practical and less costly to produce
than the multi bit converters. Low bit converters do not need a precise adjustment
of the individual bits, especially the most significant bit (MSB) as is the case
with multi bit converters. If the MSB is not precisely adjusted, the resulting
sound wave is distorted. It is clear that precision multi bit converters are expensive.
Initially Multi bit converters kept their place in stationary quality CD-players
and DAT recorders in the studios and in homes.
PULSE
DENSITY
There are two version of low bit conversion techniques. Pioneer
and Matsushita chose the so called Pulse Width Modulation. The total music
signal of 16 bits is being contained in a quantity of a few bits, in fact 3.5
or 5 bits, which do not ask for precise adjustment of any bit.
Philips developed
their own conversion based on Pulse Density Modulation. There the complete
signal is contained in just 1 bit. These manufacturers began to incorporate these
cheaper converters also in machines for home audio.
These conversion techniques
showed an improved wave form at very low recording levels where multibit converters
show a ladder instead of a sine wave.
However the improved sine wave generated
by a low bit converter (be it based on Pulse Width Modulation or Pulse Density
Modulation) is not a thin fine line as it normally should be the case, but the
wave is a thick, somewhat woolly sine wave which indicates that some manipulation
of the data and a high oversampling rate have been applied. It shows that noise
is generated
which is added to the signal and does not stem from the pure sine wave or the
recorded sound.
FILTER
When choosing a sampling frequency of 44.1 kHz., only the frequency band of 0
to 22.05 kHz. can be used because this band is mirrored by the band above the
22.05 kHz., namely the band between the frequencies from 22.05 kHz. to 44.1 kHz.
This means that each and every frequency under 22.05 kHz. has an alias in the
frequency band above 22.05 kHz. Hence the term aliasing.
It is important
that the band of 22.050 kHz. to 44.1 kHz. does not interfere with the actual audio
band. In order to make this interference impossible, a steep filter (brick wall
filter) must be used. No aliasing may occur.
By applying an oversampling
rate of four times (or a multiple of four times), the brick wall filter is not
necessary to act at at 22.050 Hz., the frequency at which the filtering normally
should take place. Because of the oversampling, a less steep filter can be applied,
say 6 dB at 22.5 kHz.. Now not only a wider frequency band is suggested, but also
the phase of the signal has improved. The fine detail which is a trademark of
a real wide frequency band is however missing.
DIFFERENCES
There are technical differences between the precise multibit converter and
the low bit converter. These differences are audible too. Listening to a recording
of a symphony orchestra or an ensemble converted by a multibit decoder the space
between the instrumentalists is apparent. One can, as it were, see the stage floor.
Denon, Accuphase and other high-end brands like Mark Levinson, Theta, PS Audio,
applied precise multibit conversion. Multibit converters have to be adjusted very
precise in order to keep the harmonic distortion very low and to achieve a linear
frequency response. Both are related to the accurate adjustment of the bits. The
technical designers began applying oversampling in multibit converters almost
from the day the CD was introduced. So they applied it in multi bit converters
and they chose an oversampling rate of 4x, later 8x, 16x, 32x and much later 64x.
The amount of oversampling determines the precision of the signal. Personally
I find 8 times oversampling an optimum for the ear. In this way the sampling frequency
is transposed to 352.800 Hz. An oversampling of 16 is to my ears the limit. Higher
rates of 32 (Wadia), 64 (Krell) and of course 256 times make an audible difference
and deliver a less "natural" quality. The higher the oversampling rate,
the more small side effects of the conversion are being multiplied. To sensitive
ears they may become audible and need to be filtered out.
MULTIBIT
AND MULTIBIT
Listening to a musical signal that has been converted by
a precise multibit decoder "shows" that the instrumentalists are seated
in front of each other, next to each other and behind each other, which means
they are seated at random so to say as on the stage. When using a multibit converter
which has noise shaping (as with players of Philips and Marantz) it seems as if
one looks over the heads of the instrumentalists. The space in between the instrumentalists
is less evident. I personally noticed many times that a low bit converter
places the musicians as if seated neatly in half a circle which is not the reality.
On top of that
the attack in the musical passages with strong dynamics and complex structures,
do sound a grade weaker, they are spread out, so to speak. Through a high rate
of oversampling, the highest frequencies sound friendlier to the ear, but are
less chiseled and less "clean". The advantage of the low bit concept however is
that the nasty ringing of a more or less steep filter is being avoided and a better
transient response is suggested.
FURTHER
DEVELOPMENTS
When in 1994 Pioneer Electronics introduced their Wide Band DAT recorder
with a sampling frequency of 96 kHz. and 16 bit quantization, Philips
technicians were immediately interested. After the press conference
in the Pioneer headquarters in Almere (Netherlands), the machine was
sent to Eindhoven at once. The engineers' interest was brought about
by the fact that an audio band from 2 Hz. to 44.000 Hz. was established
in this DAT recorder. That was exceptional as normally 24 kHz. would
be the upper limit if 48 kHz. sampling frequency was selected. Pioneer
had used 2 Philips bitstream (1-bit) converters based on 16 bit quantization
and a sampling frequency of 48 kHz. They worked together. The application
of a very expensive 32 bit converter (as was already used in the Mitsubishi
Digital Tape Recorder in the nineteen eighties), was being avoided.
Using bitstream converters in this DAT recorder opened new and inexpensive
possibilities. Nevertheless, the fact that there was noise shaping could
not be denied when listening to the Japanese train passing by and the
birds chirping in the trees. This format was of course only suitable
if recordings were made from analogous sources (live recording, Lp transfers).
Transferring CD's to the tape in this Pioneer DAT in Wide Range mode
was not possible as there was no conversion.
SUPER
AUDIO CD
The design of high speed processors made it possible to develop
the low bit conversion technique further. The result was Direct Stream Digital
(DSD) which is the basis of the Super Audio CD.
In Positive Feedback (Vol.
8, No. 2) designer Ed Meitner is being interviewed by Mike Pappas. Meitner was
involved in the implementation of the Super Audio CD. He accuses those technicians
of the early nineteen eighties who did not listen carefully enough to the new
CD format because they were completely caught by the novelty of the digital technique.
Since then they have become so familiar with the "simple" format of 44.1 kHz.
and 16 bit that they now do not see the importance of the high resolution which
DSD brings about.
Furthermore Meitner states the well known adagio that "less" is always better
than "more". In this respect he talks about circuits and buffer amps in players
that should have a discrete layout as a result of the high energies which are
generated by the vast data stream. Integrated circuits and certain operational
amplifiers cannot deal with these high energies, he says.
However if one sees the complexity and the manipulation of the signal that takes
place in the converter of the Super Audio CD one must conclude that Meitner contradicts
himself. It is not less but more.
HIGH
END?
In defending the Direct Stream Digital of Super Audio CD Meitner
says in the telephone conversation with Pappas that some high-end manufacturers
use low bit converters in their very costly DA-Converters. So why should not they
use Direct Stream Digital?
The use of low bit converters in high-end players
is of course an idiocy. Many times we have witnessed demonstrations with DA
Converters from Audio Research and Threshold at the time. When I asked the demonstrator
if it was possible to connect a multi bit DAC, in most cases they were able to
produce such a component. For all listeners it was evident that the five times
more expensive low bitters could not match the quality of the multi bit DAC.
So the use of low bit converters in high end machines and separates is not an
argument for anyone to abolish PCM and multibit conversion and switch to DSD and
low bit conversion. On the contrary! Furthermore the use of 1-bit conversion in
expensive components (Threshold, Audio Research) is not proof that it provides
a better outcome of the treatment of the data compared to what a multibit converter
does. Just listen to the Accuphase technology.
Personally I swear to precise
multibit converters as applied by Denon and high-end manufacturers like Accuphase,
Theta, Enlightend Audio, etc. And personally I would have given my preference
to a Super Audio CD if the technique would have been based on Puls Code Modulation
but with a high sampling frequency and a high level of quantization with precise
multi bit conversion. As said earlier, Mitsubishi in the nineteen eighties, already
had a digital reel to reel recorder and AD and DA converters with a sampling frequency
of 96 kHz. and 32 bit quantization. Today even higher frequencies and bit rates
would be feasible.
All this does not mean that one would not see the ingenuity of the DSD as everybody
marveled at the technique of the old CD format.
GIGANTIC
Through oversampling the conversion frequency of the Super Audio CD is 2.8224
MHz. as stated by the data sheets Such a high frequency gives a better resolution
if compared to the current PCM formats. Because of the fact that SACD uses DSD
and not PCM, the frequency band is not half of the 2.8224 MHz. but extends to
just over 100 kHz. (110.25 kHz. I suspect, no accurate frequency is mentioned
by Sony and Philips). This suggests that the oversampling rate is 256 times.
The designers of DSD bitstream state that it has two more advantages: a low harmonic
distortion and a perfect linearity.
But also regarding these two aspects, the precise multibit converters do show
an extremely high performance. The ingenious technology of Accuphase regarding
the current CD format for example does not only show this when measuring, but
it lets you hear it in reality and very clearly. And above all you do not hear
the noiseshaping filtering.
ACOUSTIC
INSTRUMENTS
Most technicians and producers are experts who are used to
dealing with pop music and digital musical instruments. The only test however
by which a converter can be really evaluated on its merits is the recording and
reproduction of acoustic instruments. When did you last visit a symphony concert?
If the sound character of the live performance of the symphony orchestra still
resonates in your ears, then you know that SACD tries to fool your ears.
R.A.B. Page first
published September 2000
