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
