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The Super Audio CD (SACD) - DSD - DXD


Considerations and Personal Preferences

SACD: No PCM but The Sound of Oversampling / Upsampling and Noise Shaping

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:

    1. a high resolution of every frequency in the audio band,
    2. an extended (practically unlimited) frequency band for upper harmonics, and
    3. a very good transient response,
    4. good dynamics,
    5. an extremely low noise level, and
    6. 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.

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.

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.


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.

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

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.

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.

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.

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.

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.

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.

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.

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

Addendum: Direct eXtreme Digital

SACD is a high capacity storage medium. DSD (Direct Stream Digital) is the format in which the data are stored on the SACD.
DSD is a 1-bit format, a so called pulse density format and definitely not a PCM format (Pulse Code Modulation).
DSD is suitable for storing high resolution analog tape recordings. Originally the DSD format (later to be used in SACD players), was developed for the storage of old analog tape recordings.
The signal is transferred without intervention into the DSD format: Direct Stream Digital.

DSD has two negative aspect:

1. Noise shaping is applied.
This deteriorates the pulse. The filter of the noise shaping is hanging like the sword of Damocles over the music signal.

Accuphase is well aware of this problem. They devised a new technology called MDSD. Its task is to reduce noise during playback of Super Audio CD (SA-CD) sources. The signal recorded on the Super Audio CD is a 2.822 MHZ/1 bit DSD signal. It has a very wide frequency response and dynamic range, but by principle it contains significant quantization noise components outside the range of human hearing, mainly in the region above 100 kHz. This noise however is interfering with the musical signal. In the MDSD technique, the 1-bit DSD signal from the input is first upsampled by a factor of two, from 2.822 to 5.644 MHz. thus placing the noise even higher in the frequency band. Now the 1 bit signal is shifted progressively in increments of 177 nanoseconds and these signals are then treated by 8 D/A converters per channel for conversion into analog sound. (See the technical explanation on the Accuphase website by checking the DC-801 SACD Player.)

2. Editing is impossible.
A digital format should have at least 2 bits to be able to edit the signal. DSD is a one bit encoding/decoding format, which means that it is not possible to edit in DSD. Only in a pulse width low bit format editing is possible.

In order to use the full benefit of the SACD with its Direct Stream Digital high resolution, and to have the possibility of editing the recording, it is of course necessary to make recordings with a high resolution medium like the analog tape recorder or any high resolution PCM (multibit) format.
After editing, the recording can be transferred to DSD and can be stored on the SACD.

But what high resolution PCM recording format? The format of the Compact Disc with its 16 bit and 44.1 kHz. or the 48 kHz. of the digital audio recorder (DAT) can hardly be called high resolution formats. Nor is the Soundstream digital technique which has 50 kHz. sampling frequency.

If editing is not possible in a 1-bit (pulse density) format, it is necessary to develop a format in which it is possible. Now a new recording format has been developed: DXD, Direct eXtreme Digital.
DXD is a format in which editing is possible. It has a higher sampling
frequency and a higher resolution than the early digital format of 44.1 kHz. sampling frequency and 16 bit.
DXD is the low bit, pulse width format, which uses 5 bits.
The advantage is not only the possibility of editing. It also needs only about half the level of noise shaping which the DSD recording system needs.
The use of Direct eXtreme Digital means that the pulse has also improved. (You can imagine what the quality would be when working in a 16, 24, 32 or 64 bit format which do not need noiseshaping at all and give maximum pulse.)

Nevertheless the introduction of DXD seems to be good news. Now any recording made in whatever format can be converted to DXD and after editing the signal in DXD, it can be converted to DSD and stored on the SACD. The use of DXD means better sound coming from SACD.

R.A.B. December 2005

See the pdf documents for all the info in this page http://www.digitalaudio.dk/page1193.aspx?q=dxd

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Audio&Music Bulletin - Rudolf A. Bruil, Editor - Copyright 1998-2008 by Rudolf A. Bruil and co-authors