DSD, WDSD and DXD

[Sinewave]

We think it’s a big step forward, offering far better impulse response that previous PCM, avoiding the filtering problems of previous PCM and better out of band noise performance than DSD. We think this is the future of digital audio.

Yeah but a DXD recorder / editor is way beyond most of us here , so like DSD it will be around and no one will be able to afford to use it.

[YGoh]

Great intro. Peter. Thank you

- From your experience, how is the conversion from DSD 64fs (or 128fs) to DXD (for processing) in the DAW? Is it very 'transparent' or fo we loose any frequency components?

- What are the filtering slope for DXD and 384kHz PCM?

- Why is it that 1 bit processing impossible?

- Is Sony e-chip (8bits 64fs) a form of pure DSD processing, and better format to convert from DSD 1bit 64fs?

[YGoh]

Peter,

- The front end of DAD's A/D converter is delta-sigma modulation (1 bit), yes?

- Then the signal is being decimated, yes?

- How is this compare to the old high bits A/D converter?

[Peter Scheelke]

Originally Posted by YGoh

Great intro. Peter. Thank you

- From your experience, how is the conversion from DSD 64fs (or 128fs) to DXD (for processing) in the DAW? Is it very 'transparent' or fo we loose any frequency components?

- What are the filtering slope for DXD and 384kHz PCM?

- Why is it that 1 bit processing impossible?

- Is Sony e-chip (8bits 64fs) a form of pure DSD processing, and better format to convert from DSD 1bit 64fs?

Originally Posted by YGoh

Peter,

- The front end of DAD's A/D converter is delta-sigma modulation (1 bit), yes?

- Then the signal is being decimated, yes?

- How is this compare to the old high bits A/D converter?

It is much better to record directly in DXD format. The principle of the DAD AD converter is to sample in 5-bit at 128 times the sample frequency. When sampling is based on 44.1 kHz the sample rate of our delta/sigma front end is 5.6448 MHz. This gives a much better and accurate signal, than if it was only derived from a 1-bit 64 og 128 fs signal as DSD.
If the media is SACD SRC to DSD is only needed once after all editing has been done.

For DXD our filter start very gentle at 125 KHz. At 200 KHz the filter attenuates about 12 db.

The simple answer to the question about 1 bit processing is that the delta/sigma modulated 1 bit signal is giving the information if the sample is one higher or lower than the previous. Further more to process at only one bit does not give any room for calculations, since any digital processing will have +/-1bit accuracy. Therefore the delta sigma signal has to be decimated to so a called multi-bit signal. DXD is indeed such a signal, but other formats are used by other vendors.

The problem is not so much to convert from DSD to any multi bit format nor the Sony e-chip (8bits 64fs). The signal deteriation appears when the multi bit signal has to be re modulated back to DSD. Then the characteristic noise of the 1-bit delta/sigma modulator again is “infected” on the signal making the out of band noise worth than it originally was on the DSD signal. When the source signal has a higher resolution (like 5bit) only the last modulation noise will affect the signal.

Best regards,

Peter

[Peter Scheelke]

In the figures below the relationship between sample rate, impulse responses, noise performances and frequency response from the A/D converter in our AX24 can be seen.

It is clear that DSD has the very best impulse response and band wide but significant more “out of band” noise than PCM and DXD. The impulse response of DXD is still good and the noise performance is better than DSD64fs.



Everybody who has ever compared DSD64fs, DSD128fs, DXD and 384 KHz to analog would probably agree, that all these formats are sounding much more transparent than PCM between 44.1 and 192 KHz. The main advantage of high sampling rate is however not the wider frequency band itself since we can not hear frequencies above 20 KHz.
High sampling rates is mainly about timing due to improved impulse response.

The front-end of our AX24 DSD/DXD converter is a 5-bit delta-sigma modulator running at 128fs. (5.6448 MHz). A multi bit A/D front-end is much more efficient than a 1 bit front-end.

I our implementation I would say that DXD/384 KHz is the most transparent format. Very close to DXD comes DSD128fs and then we have DSD64fs. There is a quite a jump between DSD64fs and 192 KHz.

If the media is SACD and editing is not needed (except for cross fades) it is probably the best solution to record in Pure DSD, since any sample rate conversion has an impact on the sound.

If editing is needed I would recommend DXD/384 KHz. If the media is SACD you will only need to convert to DSD once, if the media is something else, you will have the benefit of a higher resolution in the production phase. Also EQ, dynamic and reverb are working much better at higher sampling rates.

Best regards,

Peter

[Brad Lunde]

Originally Posted by Sinewave

Yeah but a DXD recorder / editor is way beyond most of us here , so like DSD it will be around and no one will be able to afford to use it.

I don't know, maybe Sony will decide to build a $399 DSDman portable player and kick the ipods butt. I think DXD engines in consoles are not far away. Storage has become far less of an issue as it was 5 years ago. Better/faster internet networks are being planned that could handle DXD transmission. Whether its DSD, DXD or 384K- better digital audio WILL come.
Brad

[YGoh]

Peter,

- When do you think we'll see 24bits 768 kHz PCM? Besides the obvious high data rate and storage, what are the difficulty that is associated with it?

[videoteque]

Originally Posted by YGoh

Peter,

- When do you think we'll see 24bits 768 kHz PCM? Besides the obvious high data rate and storage, what are the difficulty that is associated with it?

Hear the diference of MP3s and 24bits/768khz if music keeps been recorded and mastered as it is nowadays. Not to forget, most people's stereos (that suck!) but have a lot of different colors LEDs...

[YGoh]

Peter, I read this post about DSD's inherent noise from another forum (which I thought is really good and technical, PSW Recording Forums: Reason In Audio => Digidesign and DSD). Which of the following six points are not true?

1.) Along the lines of the research of Vanderkooy and Lipshitz, 1 bit is insufficient to avoid a stochastic breakdown either in direct recording or in noise-shaped results. Not enough random behavior can be added to the system - nor can the signal itself contain random behavior of significant enough amplitude - to decouple the quantization error in the system from the data. As such, the quantization error does not manifest itself as "noise" but rather as a form of "distortion," one that typically shows up if cyclical behavior is allowed to build in the feedback loop of the noise-shaping algorithm. This most often occurs with very low frequency material - especially DC. The fact that it is most prone to exposure with DC does not meant that this distortion is not present or audible under normal playing conditions. Because the noise-shaping algorithms have specific filter shapes the specifications that we get in the market generally highlight these filter shapes and not the worst-case scenario that occurs upon stochastic breakdown. This problem is often referred to as the little "birdies" that you hear when you record material at very low amplitude. You can force this behavior simply by putting a very slight amount of DC offset into the converters.

Other noise-shaping algorithms can avoid this simply by adding random behavior in the feedback loop. Again, however, 1 bit simply does not provide adequate room to add sufficient random behavior in the feedback loop, either. This problem has followed our industry ever since the 1 bit converter was publicly released to our industry. Fortunately, this problem was overcome through use of a random alignment of multiple 1 bit converters, so that 5 of them (for example) can be stacked up and randomly varied, providing enough data that the cyclical action can be overcome with the random varying of the 1 bit converters inside the box. DSD converters employ this technique, too! The problem is that they can't then reduce the material to only one bit without the problems described above. Meanwhile, multi-bit PCM systems have finally gotten away from that problem.

2.) Along with this issue is the increase in dynamic range from multi-bit systems that stem from multi-bit (multi-level) high frequency initial conversion processes. 1 bit systems are prone to lower dynamic range as a consequence of the shaping used to reduce the data to a single bit.

3.) SACD systems produce significant amounts of high frequency noise and therefore require (well, strongly recommend) outboard, analog filters to roll off the excess distortion (noise) above the audible frequency. These filters add phase-shift of audible amounts within the audible range. This is why the same process in multi-bit PCM converters use linear-phase filters. Those filters can avoid phase shift of audible frequencies. Unfortunately, DSD is converted to analog before linear phase filters can be utilized, so phase-shifting filters must be used.

4.) Not all of the excess noise is filtered, as the filters have a roll-off. Excess distortion (noise) above the audible band that is not filtered creates non-linearities in amplifiers and speakers that are not designed to handle the frequency response of that noise, and at that significant amplitude. Tweeters, when pressed with high frequency noise, heat up and become non-linear, causing distortion of other frequencies pressed into them - very much akin to the distortion caused by the saturation of analog tape.

5.) The SACD scarlet book format has a clever and necessary trick in place to prevent "modulator overload," a problem wherein excessively high amplitude data that can break your external gear is not allowed into the system. The method essentially forces the disk to only contain "legal" data - that data that can be represented within the amplitude response of the system. This type of safeguard is NOT in place in multi-bit PCM formats. Multi-bit PCM delivery formats actually allow you to break the boundaries of what the system is designed to reconstruct, allowing distortion. I wrote a paper on this for Trillium Lane Labs which can be downloaded at my website entitled "The Consequences of Traditional Digital Peak Meters." As I mention in the paper, this problem is most noticeable (or most pressing) in situations where significant amounts of compression are used. The less compression used the less likely the multi-bit PCM system creates this form of distortion. SACD does not allow this form of distortion by putting a safeguard in against modulator overload, so as compression is added to try to increase the perceived loudness of material, the system requires that the user continue to reduce the overall gain - completely counterproductive to the desired effect. The result is that many of the mastering engineers are finally getting the opportunity to add more dynamics to the music by compressing it less, for essentially, add dynamics or don't and the "perceived loudness" will be the same. So better to make it have dynamics! The net result is that this changes the approach mastering engineers take and prevents the ability to have fair and balanced tests between SACD releases of DSD data and the equivalent releases of multi-bit PCM data. When comparing SACD remasters with PCM original releases the SACD remasters are often said to have more "life" and other adjectives that are easily parlayable to increases in dynamics performance.

6.) The inability (or extreme difficulty) in assembling a test which overcomes these very identifiable, known, and tested issues in order to create a listening test which only tests other issues continues to exacerbate the problem by allowing non-scientifically controlled listening tests on the subject to dominate market impressions, and thus bias, and thus the effect on future tests. Listener bias in non-double blind tests (such as ABX tests) often creates up to 30% disparity in the results, so the continued exacerbation of bad information continues to drive biased testing which has an exponential effect through generations of tests.

[Peter Scheelke]

Originally Posted by YGoh

Peter,

- When do you think we'll see 24bits 768 kHz PCM? Besides the obvious high data rate and storage, what are the difficulty that is associated with it?

I think it will take years before anyone moves to 768 KHz. A higher sampling rate than DXD/384 KHz should be justified in listening tests. A main benefit of DXD is the very good impulse response due to the very relaxed anti aliasing filter. Going to 768 KHz will not improve the impulse response much.
DSD64fs has a better inpulse responce and wider frequincy band than DXD, however DXD is still soundig more transperent than DSD64fs.

Originally Posted by videoteque

Hear the diference of MP3s and 24bits/768khz if music keeps been recorded and mastered as it is nowadays. Not to forget, most people's stereos (that suck!) but have a lot of different colors LEDs...

Yes, but MP3 will develop in the years to come. Today you can have 20,000 songs stored in an I-pod. What should Apple do next? 200,000 songs? No. I think they will improve the quality.
The new MP3 algorithm from Fraunhofer supports192 KHz all the way up to linear (no compression).
There is in my opinion 3 reasons for the success of MP3.

• Storages. Yes, however hard disk space has become cheap.
• Fast downloads. Yes, but the internet will become much faster in the years to come.
• File sharing. Today you also have “Flag” for pure 48/96 KHz, which can be shared as well.

[YGoh]

Peter,

Even though DSD has better impulse response due to the very high Nyquist cut-off, the DXD is more transparent because of the inherent distortion that DSD has, which are:

1.) The impossibility to add random noise to a 1 bit system, '...1 bit is insufficient to avoid a stochastic breakdown either in direct recording or in noise-shaped results. Not enough random behavior can be added to the system - nor can the signal itself contain random behavior of significant enough amplitude - to decouple the quantization error in the system from the data. As such, the quantization error does not manifest itself as "noise" but rather as a form of "distortion,"...'

2.) '...Not all of the excess noise is filtered, as the filters have a roll-off. Excess distortion (noise) above the audible band that is not filtered creates non-linearities in amplifiers and speakers that are not designed to handle the frequency response of that noise, and at that significant amplitude. Tweeters, when pressed with high frequency noise, heat up and become non-linear, causing distortion of other frequencies pressed into them...'

Is this true? and is this why most claim that DSD sound more like 'analog'?

[Bob Olhsson]

Real world performance is all about the best implementation as opposed to any theoretical advantages. Vanderkoy and Lipshitz show no signs of developing a wayback machine so that we can go back and re-record everything after they've perfected the PCM converter.

We have to make do with the best sounding technology we can get our hands on. All of these theoretical arguments smack of people having an agenda other than great sound.

[videoteque]

All of these theoretical arguments smack of people having an agenda other than great sound.

The problem is most people's stereo is bad enough that you can't even distinguish an 128k MP3 from a a good SACD. In this scenario, higher resolution systems are a true none sense, except for archival and audiophile(file!) recording!

[Sinewave]

Originally Posted by videoteque

The problem is most people's stereo is bad enough that you can't even distinguish an 128k MP3 from a a good SACD. In this scenario, higher resolution systems are a true none sense, except for archival and audiophile(file!) recording!

Even though most people have Horrible playback systems that is no excuse to not use the best recording technologies that are available or that one can afford.
We are in the business of composing , arranging, producing and engineering music, let the public decide on their poisonous playback.

[Sounds Great]

Originally Posted by videoteque

The problem is most people's stereo is bad enough that you can't even distinguish an 128k MP3 from a a good SACD.

I could easily determine the difference on any system including cheap computer speakers, of course you won't get SACD out of your computer.

Most people have at least adequate headphones where the difference is plain as day.

[Bob Olhsson]

Originally Posted by videoteque

The problem is most people's stereo is bad enough that you can't even distinguish an 128k MP3 from a a good SACD. In this scenario, higher resolution systems are a true none sense, except for archival and audiophile(file!) recording!

Except that scenario has absolutely nothing to do with the real world of exposing and selling records where one person's impression frequently makes or breaks a title's commercial success. Poor audio quality has frequently been the deal breaker when all else was equal. Every recording is an integral part of the artist's resume!

[Peter Scheelke]

Originally Posted by YGoh

Peter,

Even though DSD has better impulse response due to the very high Nyquist cut-off, the DXD is more transparent because of the inherent distortion that DSD has, which are:

1.) The impossibility to add random noise to a 1 bit system, '...1 bit is insufficient to avoid a stochastic breakdown either in direct recording or in noise-shaped results. Not enough random behavior can be added to the system - nor can the signal itself contain random behavior of significant enough amplitude - to decouple the quantization error in the system from the data. As such, the quantization error does not manifest itself as "noise" but rather as a form of "distortion,"...'

2.) '...Not all of the excess noise is filtered, as the filters have a roll-off. Excess distortion (noise) above the audible band that is not filtered creates non-linearities in amplifiers and speakers that are not designed to handle the frequency response of that noise, and at that significant amplitude. Tweeters, when pressed with high frequency noise, heat up and become non-linear, causing distortion of other frequencies pressed into them...'

Is this true? and is this why most claim that DSD sound more like 'analog'?

Yes, but DSD dos not have inherent distortion.. The reason while we think DXD is more transparent is that is has much less out of band noise. However very superior DSD “direct-cut” recordings has indeed been made, so we don’t think that the basis noise of DSD is causing serious problems.

1) Sorry but this is really not very correct !! First it has to be understood that all modern converter chips/systems uses a delta/signa modulator when sampling the audio. Year ago this was normally one bit, today it is mostly 5 bit. The sampling is performed at 64 or 128 times the basis sample frequency being either 44.1 or 48 khz. So the “raw” sampled data is always in this format. It is only a matter of digital representation to present the signa as PCM (multi bit) or DSD (delta/sigma data). The principle difference between PCM and DSD is that PCM represent a sample exact in time with the accuracy of the number of bits (e.g. 16 or 24 bit). DSD is a signal that represents the difference to the sample before, which again refers to the sample before… etc. the accuracy of this “difference” relates the number of bits (e.g. 1 or 5 bit ).

There is not involved any kind of distortion or stochastic breakdown in these processes as long as they are performed mathematically correct.

2) The 1 bit delta/sigma signal – DSD – has an out of band noise depending of the quality/accuracy of the specific modulator. As long at this noise is complying to the specification of the scarlet book, the problems that you refer to should not be a major issue. However if the DSD signal is decimated into multi-bit for editing, and the afterwards re-modulated into DSD additional noise is added, and such a signal can not comply to the scarlet book, and therefore it has to be filtered etc. and the the “digital” quality of the signal has suffered - which indeed is audible.

3) DSD audio material – assuming that it is properly cared for – does indeed sound more analog, as it is also the case with DXD. The reason is simple. The bandwidth is higher, and the aliasing and reconstruction filters can be much more “relaxed” this gives a better pulse response, faster attach and less ringing.

[Kees de Visser]

Originally Posted by Peter Scheelke

Year ago this was normally one bit, today it is mostly 5 bit.

Peter, AFAIK the DSD (64FS) format was established when most modulators were 1 bit. I can understand that there is some theoretical charm in recording the ADC modulator's raw output. Since most modulators are 5 bit these days, wouldn't it make sense to record all of these bits ?

[videoteque]

Don't misunderstand me! I am the first one who is FOR good audio, good recordings and good playback.

I don't understand how artists, the ones who are really famous, don't complain about over-compression. And all people involved in audio shoud try to up the awareness about having a good stereo reproduction. I have seen musicians/sound techs that have stereos that were a joke...???

How can someone achive great sound if they never heard it???

Sorry for my off-topic-rant!

[YGoh]

Originally Posted by Peter Scheelke

.........DSD is a signal that represents the difference to the sample before, which again refers to the sample before… etc. the accuracy of this “difference” relates the number of bits (e.g. 1 or 5 bit )......

- So if the difference between the current sample signal to the previous sample signal is positive, then, with 1 bit Delta/Sigma, the DSD signal will be '1'. If the difference between current and previous signal is negative, then it will be a '0'. Is this correct?

- So with the more bits to represent difference between current and previous sample signal, the more accurate you can represent the difference, is this true?

- If so, then with 1 bit, as oppose to 5 bits, then there still be more quantization error. Therefore, this error is not de-correlated (not random) from the signal, so this is distortion, right?

[Peter Scheelke]

Originally Posted by Kees de Visser

Peter, AFAIK the DSD (64FS) format was established when most modulators were 1 bit. I can understand that there is some theoretical charm in recording the ADC modulator's raw output. Since most modulators are 5 bit these days, wouldn't it make sense to record all of these bits ?

Originally Posted by YGoh

- So if the difference between the current sample signal to the previous sample signal is positive, then, with 1 bit Delta/Sigma, the DSD signal will be '1'. If the difference between current and previous signal is negative, then it will be a '0'. Is this correct?

- So with the more bits to represent difference between current and previous sample signal, the more accurate you can represent the difference, is this true?

- If so, then with 1 bit, as oppose to 5 bits, then there still be more quantization error. Therefore, this error is not de-correlated (not random) from the signal, so this is distortion, right?

1)If the only purpose is to capture the audio at its best and no editing is needed it would make theoretically sense to record the output of the 5 bit modulator. However, no DAW or digital interfaces can handle this huge amount of bits, such a format has not been justified by listening test and such a format still needs conversion to mulit bit if editing is needed.

2)Yes, a DSD system is using the previous sample and more bits in a DSD system will increase the precision. The bits in a PCM system are a digital representation of the actual amplitude. A PCM system is therefore needed for editing.

3)Yes.

4)No, since the difference between the analog amplitude and the “grid” of the DSD is random the error is noise.

[YGoh]

Peter Scheelke,

1.) Can the Merging Tech.'s Sphynx2 also do the 384kHz A/D and D/A, just like the AX24? because Sphynx2 is basically AX24, yes

2.) So the Pyramix also process audio, mix and edit at 384kHz audio data, or do they downsample to 192kHz or lower sampling rate first?

[beyarecords]

Originally Posted by YGoh

1.) Can the Merging Tech.'s Sphynx2 also do the 384kHz A/D and D/A, just like the AX24? because Sphynx2 is basically AX24, yes

The Merging Sphynx2 is an OEM DAD AX24 and differs from the DAD AX24 only in the branding on the casing.

Originally Posted by YGoh

2.) So the Pyramix also process audio, mix and edit at 384kHz audio data, or do they downsample to 192kHz or lower sampling rate first?

Algorthmix Plugins within Pyramix operate at up to 384KHZ, so yes Pyramix can process, mix and edit audio at 384KHz without the need to downsample.

[Curve Dominant]

Originally Posted by Peter Scheelke

DSD dos not have inherent distortion

Peter,

How is it possible that a 1-bit signal, impossible as it is to dither (because one needs at least 3 bits to properly dither: 2LSB for TPDF dither + 1 bit for passing audio), does not have inherent distortion?

Admittedly I have been absent from these discussions for awhile, so I'm now anxious to learn if that rabbit has been successfully pulled from that hat while I've been away.

-Eric Vincent @ Studio Curve Dominant

[zenmastering]

Dithering would be necessary if you needed to process the signal...and that isn't what Peter's talking about.


Dither is also used to stabilize some DSD sigma-delta modulator topologies


Read This

Graemme

[Ari-M.]

Originally Posted by videoteque

The problem is most people's stereo is bad enough that you can't even distinguish an 128k MP3 from a a good SACD. In this scenario, higher resolution systems are a true none sense, except for archival and audiophile(file!) recording!

huh??? I can hear the difference over the worst of playback systems....at the end of the day does any of this technology help the consumer make a better decision about the "best" format....NO....sink or swim is the old addage....

[MA Recordings]

Originally Posted by MBishopSFX

I work almost exclusively with DSD on an everyday basis and have since 1996 and have worked with every DSD recording system and workstation made so far.

There is a HUGE difference between how SADiE and Pyramix handle DSD editing. Same for Sonoma vs Pyramix.

All start with a 1-bit DSD source. All can deliver a final 1-bit DSD "master." How they get there is a different story, though.

The Pyramix, the instant you put the system in edit/mix mode, converts the DSD source to 24bit,325.8kHz pcm for all editing and then re-converts back to DSD for the output to DACs or mastering. All DSD audio, from beginning to end of what you're working on, has gone through the conversion/re-conversion process. This is NOT just simple math being performed. The audio goes through internal pcm and sigma/delta converters for the process.

SADiE (or Sonoma) keeps the DSD source as DSD (no conversion) except at the actual edit crossfade, where an insert occurs the length of the crossfade of 8bit,2.8Mhz pcm and back to DSD. This conversion occurs ONLY at the crossfade. All DSD source on either side of the edit is untouched as long as the "bypass mixer" button is pushed (how I almost ALWAYS run the SADiE DSD). I defy anyone to hear the conversion effect for the duration of an edit crossfade! However, even if the mixer is inserted, the audio integrity is more transparent than that of the pyramix to MY ears. There is no low-pass filtering used or needed. The "annex" DSD metering verifies there is no increase in UHF noise.

(Here's where I pull on the flame-proof clothing ) In MY experience, there is a big difference in how these system sound as well. The Pyramix, as soon as the DSD source hits the 325 mHz pcm mixer (that's every mode other than source recording), changes the sound in an unacceptable manner, IMO. Philips and Pyramix claim otherwise, but I have had 6 very experienced DSD editors and skilled listeners here pick out the Pyramix vs SADiE or Sonoma audio every time. The SADiE and Sonoma keep my DSD source intact.

As with any tool in any format, taking care in how intrusive one is on the source audio will make a difference in the final outcome. The vast majority of my DSD projects can stay in native DSD except for the edits. For any processing or mixing, I come out to analog via EMM Labs dacs and record the process or mix back to DSD via EMM Labs adc. All properly-designed analog gear has no problem whatsoever with the DSD source - no low-pass filtering needed here either! This conversion to analog and back in MY experience is actually less intrusive than the Pyramix 325 mHz conversion process (I'm pulling on the second layer of flame-proof clothing...). Others may disagree (they may own a Pyramix), but this is the way I hear it!

Michael forgot to mention that the SADiE sounds better in PCM as well.......basically as close to unaltered as possible. Pyramix is like putting Tabasco Sauce on fine Sushi...... (I am writing from Japan)

MA Recordings

[DSD_Mastering]

Originally Posted by MA Recordings

Michael forgot to mention that the SADiE sounds better in PCM as well.......basically as close to unaltered as possible. Pyramix is like putting Tabasco Sauce on fine Sushi......
MA Recordings

What?? I see your latest Caro Mitis SACD was "Recording, editing and mixing on the Pyramix system by Merging Technologies, Switzerland." That must be one HOT SACD!!

Regards,
Bruce

[hirezsqwave]

Peter,

1) If a sweot square wave is fed into the AX-24 A to D and recorded at 384 kHz, at what frequency will the recorded signal become a sine wave as stored on the Pyramix system?

2) On the output of the D to A is it normal for the morph to a sine wave to take place at 33-34 kHz? If so, is it possible to modify the LPF on playback to allow audio approaching the nyquist frequency? (The reason for this question is question number 3 below)

3) My good friend is interested in researching the sounds emitted by dolphins. An unusually wide frequency range is desired. Is it possible for the Sphinx II and Pyramix to be modified to allow for DSD 128fs recording to afford detection of high frequencies in the range of DC to 300 kHz? It is highly probable that those conducting other kinds of aquatic research would be interested in your reply. Thus, my post of this question here. Thank you for your considered response and time for these queries.

4) If the dolphins were recorded at the 384 kHz s.r., and it was desired to extract frequency shift data for musical purposes, would pitch shift and analysis plugins work at the 384 kHz s.r.? Thank you again.

[armilus]

Is DSD-Wide 8bit/64fs PDM or PCM editing format for DSD64? And if DSD-Wide is PDM, does it use any decimation filters?