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Ayre wants $1.5K for DSD'ed QB-9


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And just last night we finished the first prototype KX-5 preamplifier, a companion to the new VX-5 power amplifier (shipping since March).

 

Oh, don't tell me that! I've been trying to figure out how to afford an AX-5 and you up the ante!

 

BTW, someone on AA is wondering about the "Diamond output stage".

Roon ROCK (Roon 1.7; NUC7i3) > Ayre QB-9 Twenty > Ayre AX-5 Twenty > Thiel CS2.4SE (crossovers rebuilt with Clarity CSA and Multicap RTX caps, Mills MRA-12 resistors; ERSE and Jantzen coils; Cardas binding posts and hookup wire); Cardas and OEM power cables, interconnects, and speaker cables

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I think he left out one line

 

"Was picked on in school as a kid:)"

 

Hello Lab,

 

In school, I think there are only two kind of kids. The ones who got picked on, and the ones who did the picking. Does that mean that you were the school bully? If not, why not more compassion for this man you have never met?

 

Sincerely,

Charles Hansen

Dumb Analog Hardware Engineer
Former Transducer Designer

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Oh, don't tell me that! I've been trying to figure out how to afford an AX-5 and you up the ante!

 

BTW, someone on AA is wondering about the "Diamond output stage".

 

Hello Beetle,

 

Don't worry. You will be more than happy with the AX-5. Plus you save money on one power cord an one (long?) pair of interconnects.

 

Thanks for the pointer. I will go look.

 

Best,

Charles Hansen

Dumb Analog Hardware Engineer
Former Transducer Designer

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why not more compassion for this man you have never met?

 

Because DSD!

 

DON"T YOU GET IT, MAN?

 

DSD64, DSD128, DSD256, DSD512! You won't merely hear the ghosts of dead artists! We're talking DSD-Jesus!

Roon ROCK (Roon 1.7; NUC7i3) > Ayre QB-9 Twenty > Ayre AX-5 Twenty > Thiel CS2.4SE (crossovers rebuilt with Clarity CSA and Multicap RTX caps, Mills MRA-12 resistors; ERSE and Jantzen coils; Cardas binding posts and hookup wire); Cardas and OEM power cables, interconnects, and speaker cables

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does anyone know the list price on the new qb9 dsd model..?

What, you didn't feel like wading thru' this fiasco of a thread? With it's wildly wrong title? And substantial detour?

 

Here is the answer to your question:

http://www.computeraudiophile.com/f6-dac-digital-analog-conversion/ayre-wants-%241-5k-dsded-qb-9-a-15650/index2.html#post222829

Roon ROCK (Roon 1.7; NUC7i3) > Ayre QB-9 Twenty > Ayre AX-5 Twenty > Thiel CS2.4SE (crossovers rebuilt with Clarity CSA and Multicap RTX caps, Mills MRA-12 resistors; ERSE and Jantzen coils; Cardas binding posts and hookup wire); Cardas and OEM power cables, interconnects, and speaker cables

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Dear Wisnon,

 

d) No manipulation of the signal is possible without converting the signal to PCM. Then it needs to be reconverted to "DSD", adding even more out-of-band noise. You can't even adjust the levels or do a fade, let alone EQ or add reverb or any other effects. It is completely useless as a recording format for modern music. The Sonoma workstation converts the DSD signal to 8-bit PCM, so probably less than 1% of all SACD's ever released weren't converted to PCM at some point in their production process.

 

 

Wow Charles!! I just saw this. I respect you and i LOVE your amps. But i think you are so wrong on this issue.

First, NONE of the music done for Acoustic Sounds' Analogue Productions (other than occasional tiny crossfades and small fade in and outs) ever get converted to PCM and they are making more SACD's than anyone. So the 1% statement is wrong. There are plenty of analog to DSD SA-CD's out there.

And if we do EQ, mix, compress, or change levels in the Sonoma, then the PCM sample rate is at 2.8 million Hertz. (You don't mention this at all.) IMHO, PCM at the DSD rate still feels like DSD and not at all like PCM at lower sample rates (Even 192k PCM). I know this is a subjective statement, but we are dealing with both DSD and hi-end PCM here all the time. They are not at all the same animals. I believe that the high sample rate is VERY important. Far more so than whether or not something is DSD or PCM.

Also, it is a bit misleading to say we convert to PCM to do processing because it leads one to think we are dropping the sample rate frequency too. We are not. (When most people think of "PCM" they think 192kHz or 384kHz is the top rate).

Responding to the statement that all but 1% of SACD's go through PCM - even if it WERE true - SO WHAT? PCM at 2.8Mhz makes a great sounding record and still sounds like DSD!

As for making the choice to process in PCM at the DSD rate or not, it is like anything else in making a record. You evaluate if it is necessary to go through it. There is some loss to consider just as there is in going through ANYTHING analog or digital. If the music needs something, then you make that choice... Leave it as it is? Or process? Do it in the analog domain? Or use the DSD mixer?

And to your statement about DSD being completely useless for modern recording, we are doing multitrack music production to DSD every day at Immersive Studios next door. Charles, please just stop by and chat with the engineers next door recording modern music to DSD every day. We're just down the street!! Ask them what they think!

 

gus...

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Wow Charles!! I just saw this. I respect you and i LOVE your amps. But i think you are so wrong on this issue.

First, NONE of the music done for Acoustic Sounds' Analogue Productions (other than occasional tiny crossfades and small fade in and outs) ever get converted to PCM and they are making more SACD's than anyone. So the 1% statement is wrong. There are plenty of analog to DSD SA-CD's out there.

And if we do EQ, mix, compress, or change levels in the Sonoma, then the PCM sample rate is at 2.8 million Hertz. (You don't mention this at all.) IMHO, PCM at the DSD rate still feels like DSD and not at all like PCM at lower sample rates (Even 192k PCM). I know this is a subjective statement, but we are dealing with both DSD and hi-end PCM here all the time. They are not at all the same animals. I believe that the high sample rate is VERY important. Far more so than whether or not something is DSD or PCM.

Also, it is a bit misleading to say we convert to PCM to do processing because it leads one to think we are dropping the sample rate frequency too. We are not. (When most people think of "PCM" they think 192kHz or 384kHz is the top rate).

Responding to the statement that all but 1% of SACD's go through PCM - even if it WERE true - SO WHAT? PCM at 2.8Mhz makes a great sounding record and still sounds like DSD!

As for making the choice to process in PCM at the DSD rate or not, it is like anything else in making a record. You evaluate if it is necessary to go through it. There is some loss to consider just as there is in going through ANYTHING analog or digital. If the music needs something, then you make that choice... Leave it as it is? Or process? Do it in the analog domain? Or use the DSD mixer?

And to your statement about DSD being completely useless for modern recording, we are doing multitrack music production to DSD every day at Immersive Studios next door. Charles, please just stop by and chat with the engineers next door recording modern music to DSD every day. We're just down the street!! Ask them what they think!

 

gus...

 

Gus,

 

Gus, I love your Authoring & Mike Yach Recording on Janet Feder's "Songs With Words"!

 

Songs With Words | Janet Feder

 

Janet Feder.jpg

 

BTW, It was too hard to recording, mixing & mastering?

 

Congratulations!

 

Roch

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And if we do EQ, mix, compress, or change levels in the Sonoma, then the PCM sample rate is at 2.8 million Hertz. (You don't mention this at all.) IMHO, PCM at the DSD rate still feels like DSD and not at all like PCM at lower sample rates (Even 192k PCM). I know this is a subjective statement, but we are dealing with both DSD and hi-end PCM here all the time. They are not at all the same animals. I believe that the high sample rate is VERY important. Far more so than whether or not something is DSD or PCM.

 

Hi Gus. Thanks for your input. I've always thought that high sample rates were key, but every time I've commented about it, It starts the Nyquist debate all over again. Robert Watts of Chord thinks sample rates at a minimum 1 Mhz are necessary accurately capture transients and I believe it's true.

 

There is something I like about DSD that I don't hear with PCM. However, I do hear a bit of harshness. I'm wondering if that problem is something that has to do with some of the process used in the digital transfers which I've heard. Or a problem that will maybe be solved with DSD128 or DSD256? Keep in mind I've never heard an analog master so I have no reference to compare.

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Hi Gus. Thanks for your input. I've always thought that high sample rates were key, but every time I've commented about it, It starts the Nyquist debate all over again. Robert Watts of Chord thinks sample rates at a minimum 1 Mhz are necessary accurately capture transients and I believe it's true.

 

There is something I like about DSD that I don't hear with PCM. However, I do hear a bit of harshness. I'm wondering if that problem is something that has to do with some of the process used in the digital transfers which I've heard. Or a problem that will maybe be solved with DSD128 or DSD256? Keep in mind I've never heard an analog master so I have no reference to compare.

 

Bold is mine.

 

Recording engineer 'overload'? This is what I got as an answer from a recording engineer I admire... Then not a DSD recording issue per se!

 

Roch

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Hello Gus,

 

Thanks for your input. This thread has gone WAY off track. It originally started because we added the DoP feature to our USB DACs so that they could play DSD files. Gordon Rankin and I first talked about adding this feature when you told us about DSD-DISC at the RMAF seminar where we were on the same panel. I was pretty excited about non-copy protected (ie, no PWM or PSP). I told him that it would be very easy to "packetize" the DSD stream into "byte-size" pieces and send it over the USB connection. He got very excited also.

 

We talked about the technical detail for about 10 minutes and had the whole system figured out. Then we both said, "Wait a minute. This is stupid. The only source for DSD files would be to rip SACD's. Since it is illegal in the US to do this under the DMCA (except as a personal back up), there is no point to make a DAC for a format with no source material."

 

Since that time there are three or four VERY small companies that have a handful of releases on DSD. Some people are very excited about this. In a way I cannot blame them. The reason is that is very much like vacuum tubes -- you have to work VERY hard to make a BAD sounding DSD recording or a BAD sounding vacuum tube amplifier.

 

We studied the reason for this and it boils down to one simple fact -- the digital filters that are used in PCM. So it is ironic that turns out that DSD showed us the way to make good sounding PCM. But DSD has many problems. These problems are the least objectionable when it is used ONLY as a release format. In that case, DSD will sound MUCH better than Redbook CD.

 

But that is not a fair comparison. It simply is not fair to compare a high-resolution system with a high bit rate against Redbook CD. A better comparison is to compare against a high-resolution PCM system. Up to now, all high-res PCM systems are designed by digital engineers. The same engineers wearing blinders that developed Redbook CD now design "high-res" PCM. And so naturally they made the SAME mistakes that they made when the developed Redbook CD.

 

But by learning the lessons that DSD taught us, we can make high-res PCM with equal or better sound quality than DSD.

 

[Wow Charles!! I just saw this. I respect you and i LOVE your amps. But i think you are so wrong on this issue.

 

First, NONE of the music done for Acoustic Sounds' Analogue Productions (other than occasional tiny crossfades and small fade in and outs) ever get converted to PCM and they are making more SACD's than anyone. So the 1% statement is wrong. There are plenty of analog to DSD SA-CD's out there.

 

If I said 1%, that was taken out of context.

 

You are referring to simply transferring an analog master tape to SACD, where SACD is used only as the release format. In that case it is not a bad format EXCEPT that it requires all new hardware for both the studio and the consumer.

 

In my quote you cited, I was referring to material that was originally recorded in DSD.

 

And if we do EQ, mix, compress, or change levels in the Sonoma, then the PCM sample rate is at 2.8 million Hertz. (You don't mention this at all.) IMHO, PCM at the DSD rate still feels like DSD and not at all like PCM at lower sample rates (Even 192k PCM). I know this is a subjective statement, but we are dealing with both DSD and hi-end PCM here all the time. They are not at all the same animals. I believe that the high sample rate is VERY important. Far more so than whether or not something is DSD or PCM.

 

OK, that is fine and we are in agreement there.

 

My point is that Sony LIED and MISREPRESENTED DSD to make it seem like it was some special "magical" system and that PCM was inherently flawed. That is not true.

 

You understand that and I understand that and we both agree.

 

But the problem is that there are MILLIONS of people who have drunk the Sony Kool-Aid and believe that there is something "magical" about DSD and therefore PCM is inherently evil. But you and I both know that is not the case.

 

[NB: To all the people who do not know who "Sonoma Gus" is, he is Gus Skinas. He was involved with Sony on the SACD project from the very start. He originally was the liaison between Sony and the recording studios. He would work with the studios to help them make projects in DSD. When Sony abandoned the SACD project (because it was losing money for them, they gave Gus all of the rights to the Sonoma. The Sonoma is the Digital Audio Workstation (DAW) that is used in recording studios to perform all of the things required in a studio -- EQ, reverb, effects, et cetera. It does this by converting the DSD signal that is only 1-bit to an 8-bit PCM signal that operates at the same 64x frequency as the 1-bit DSD signal. At this high frequency, NO BRICKWALL FILTERS ARE REQUIRED, so PCM sound just as good (or better) than DSD. However, this conversion should ONLY be done one time, because when the signal is converted back to 1-bit DSD, ultrasonic noise is added. You can get away with this if you only do it once, but every time the conversion is done, more noise is added. Eventually you will have a useless signal with more noise than music. Gus knows this so he is very careful when starting with analog tape to use the Sonoma DAW as little as possible and only ONE time on each project.]

 

Also, it is a bit misleading to say we convert to PCM to do processing because it leads one to think we are dropping the sample rate frequency too. We are not. (When most people think of "PCM" they think 192kHz or 384kHz is the top rate).

 

No, Sony is the misleading one. I have never said (nor implied) that converting to PCM requires changing the sample rate. We both agree that a high sample rate is crucial for good sounding digital because that is the ONLY thing that allow us to avoid the use of steep (especially brickwall) filters. I have said many times in this thread that a brickwall filter is what kill the sound and is responsible for bad-sounding digital. With Redbook CD, there is almost no choice. By the time you get to quad-rate PCM, one can use a filter that has NO overshoot, ringing, or other transient problems. In this case it will sound BETTER than DSD.

 

This is because DSD REQUIRES a third-order analogue filter in the analog playback stage. In contrast, PROPERLY IMPLEMENTED PCM with filters that exhibit perfect transient response do not require even third-order filters in the analog stage because they do not have the ultrasonic noise that DSD has.

 

Responding to the statement that all but 1% of SACD's go through PCM - even if it WERE true - SO WHAT? PCM at 2.8Mhz makes a great sounding record and still sounds like DSD!

 

YES!!! We both agree that PCM and DSD are both capable of making GREAT sounding recording. The only requirement is to get rid of filters (especially brickwall filters) that have poor transient response. The ONLY filter that has perfect transient response is one with a rolloff of -6 dB/octave or less (first-order filter). That is why the third-order filter (-18 dB/octave) required for DSD playback will degrade the sound compared to a high-sample rate PCM recording which does not require this filter.

 

As for making the choice to process in PCM at the DSD rate or not, it is like anything else in making a record. You evaluate if it is necessary to go through it. There is some loss to consider just as there is in going through ANYTHING analog or digital. If the music needs something, then you make that choice... Leave it as it is? Or process? Do it in the analog domain? Or use the DSD mixer?

 

In my opinion, it should be done in the native format of the recording. So if you have an analog master tape and you want to release as a digital download, all of the processing should be done in analog. The FINAL step should be the conversion to digital. That will give the least amount of loss.

 

If a recording is made in DSD format, then it should be processed in the Sonoma. There will be some degradation because it is converted to PCM for processing and then back to DSD for release. HOWEVER, the amount of noise added with ONLY ONE conversion through the Sonoma is negligible. If you start with -120 dB S/N and it loses +6 or +10 dB of noise by the conversion, this is hardly audible in the real world.

 

My problem is that Sony LIED about the Sonoma and pretended that it was only a 1-bit machine and that there was something "magical" about 1 bit and that PCM is inherently flawed. So on one side of their mouth they were lying about the so-called "flaws" of PCM, and on the other side of their mouth they were lying about the fact that all of the DSD recordings (not analog transfers) were using PCM!

 

But my disagreement is not with you Gus. It is with SONY.

 

They are the ones that LIED.

They are the ones that put so much confusion into the minds of the general public. Just look at these forums. The people here know a thousand times more about digital technology than the average consumer. And yet even here most of the forum participants are confused about what is "DSD", what is PCM, what is the Sonoma, what causes bad digital sound.

 

It is all the fault of SONY, who had BILLIONS of dollars at stake, and just like politicians in the same position, LIED to confuse people and try to save their money and power.

 

And to your statement about DSD being completely useless for modern recording, we are doing multitrack music production to DSD every day at Immersive Studios next door. Charles, please just stop by and chat with the engineers next door recording modern music to DSD every day. We're just down the street!! Ask them what they think.

 

Yes, Gus, it is true. One can make an excellent sounding recording in DSD.

 

BUT to do so, one should really start with a DSD-128 (double-speed) or DSD-256 (quadruple-speed) recording equipment. And then the engineers must understand the limitations of DSD and be VERY careful how many times they run it through the Sonoma. Standard DSD should not be converted more that ONCE. If they start with special high-speed DSD, they can convert it a few times.

 

But my question is, "Why bother?"

 

In the past, all of the high-res PCM equipment used brickwall filters, and so even the best of them would have some audible disadvantages compared to DSD. But now that we have used our heads to understand WHY DSD sounds good, we can make even quad-rate PCM that sounds as good or better than DSD, AND:

 

a) Uses standard equipment.

 

b) Uses standard techniques.

 

c) Is not limited in the number of conversions that can be made.

 

d) Does not require the consumer to purchase new playback hardware

 

e) Does not require a "format war" or dual-inventory by the distributors.

 

The bottom line is that excellent results can be achieved with either format. It is easier to get good sound from DSD because Sony was smart enough to hire Andreas Koch and Ed Meitner to design all the original hardware, and because no brickwall filters are required.

 

But by implementing PCM PROPERLY (and this does NOT just mean 192/24, this means using equipment designed by people who use their brains AND their ears to get rid of the brickwall filters), then the sonic performance of DSD can be equaled or surpassed with a STANDARD format. DSD will NEVER be a standard format. You will NEVER download DSD from iTunes. You will NEVER stream DSD on Mog.

 

It is the same situation as tubes versus solid state. Any half-wit with a soldering iron can go into his basement and make a lovely sounding tubed preamplifier. But to make a solid-state preamplifier that captures the beauty and magic of music is not so easy. There are only perhaps a half-dozen solid-state preamps on the planet that can do this.

 

Most solid-state engineers are like digital engineers. They live in a world where all wires sound the same, all CD players sound the same, and just use standard, textbook techniques with NO imagination. They will just buy some cheap IC's and think to themselves, "It must be great, because it measures perfectly!" But he never notices that it sounds like dog shit.

 

And the normal digital engineer is the same. He will use PCM because that is what is in the textbook and he will use brickwall filters found in the stock cheap chips because that is what the textbooks say to do. And he will think to himself, "It must be great, because it measures perfectly!" But he never notices that it sounds like dog shit.

 

And then the people who actually LISTEN to the equipment notice that the standard PCM equipment sounds like dog shit, and they have drunk Sony's Kool-Aid and they say to themselves (or on the forums), "AHA! It must be that EVIL PCM!". But it is NOT the "evil PCM", it is the EVIL BRICKWALL FILTERS!

 

DSD sounds fine, but will always be a niche format, just like tubes sound fine. You will never see a piece of tubed gear at the Apple Store and you will never see a DSD release on the iTunes store. But it is MUCH bigger problem for software than it is for hardware.

 

There will ALWAYS be creative people that will be perfectly happy to make tubed equipment in their garage. They will be happy to have a little one-man company and make equipment that they like and if they can make a living at it, then they will be completely satisfied.

 

But for software, NO musician wants to be on some dinky little audiophile label and sell 1000 copies world-wide. Or if they do, they probably suck. The great music of the world will appeal to MANY people and the musicians that make that music want to be on MAJOR RECORD LABELS. And the major record labels of the world have already tried DSD. They only did it whn Sony BRIBED them to do it. But they still lost money on it. (If they had MADE money, they would still be doing it because the entire goal of their company is not to make music, but to make MONEY.)

 

So the reality of the situation is, whether you believe it or not, there will NEVER be a widespread variety of great musicians on available on DSD. So if you are too impatient to wait for the mainstream to adopt PROPERLY EXECUTED PCM, you can entertain yourselves for another year or two listening to third-rate artists on tiny label.

 

I am more interested in finding REAL WORLD solutions that will bring GREAT SOUND (as good or better than DSD) to the MAJOR labels so that we can all listen to great sounding music that is available even at iTunes or Mog or Spotify....

 

Musically yours,

Charles Hansen

Dumb Analog Hardware Engineer
Former Transducer Designer

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We studied the reason for this and it boils down to one simple fact -- the digital filters that are used in PCM.

 

By making leaky filters for 4x rate doesn't fix it though. ADC chips have had these "low delay" filters for ages, especially AKM chips used widely for pro-audio equipment.

 

To make it work for PCM you have to pick a filter design algorithm and sampling rate where length of the filter is at least shorter than half-wave of 20 kHz sine (preferably shorter than half wave 50 kHz sine) and has at least 120 dB attenuation at Nyquist frequency. 4x rate is not quite enough for that.

 

For DSD it's not so challenging to have it, since Nyquist frequency is already at 1.4 MHz for DSD64.

 

So as long as when I input 1 µs pulse to ADC and see either ringing longer than 25 µs or any aliases in Nyquist band I'm not happy.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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By making leaky filters for 4x rate doesn't fix it though. ADC chips have had these "low delay" filters for ages, especially AKM chips used widely for pro-audio equipment.

 

To make it work for PCM you have to pick a filter design algorithm and sampling rate where length of the filter is at least shorter than half-wave of 20 kHz sine (preferably shorter than half wave 50 kHz sine) and has at least 120 dB attenuation at Nyquist frequency. 4x rate is not quite enough for that.

 

For DSD it's not so challenging to have it, since Nyquist frequency is already at 1.4 MHz for DSD64.

 

So as long as when I input 1 µs pulse to ADC and see either ringing longer than 25 µs or any aliases in Nyquist band I'm not happy.

 

Miska,

 

Thank you for your input. However, we do not use AKM digital filters. So you have no idea what our equipment sounds like.

 

I am glad that you are "Superman" who knows what audio equipment sound like just by looking at the specification. But I am just a dumb analog hardware engineer (who spends a lot of time listening), so your criterion makes no sense to me.

 

For you an A/D converter will only sound good if it responds properly to a 1 µs pulse. This is means you want your audio A/D converter to respond properly to (approximately) 1,000,000 Hz. I know that I am just a dumb analog hardware engineer, but can you please tell me which musical instruments have harmonics at 1,000,000 Hz?

 

Or maybe you can tell me what microphones have flat frequency response to 1,000,000 Hz?

 

And then when I read more, I become even more confused. In my experience filters that ring are perhaps the largest problem with the sound quality of digital audio. But you think that it is OK if the A/D converter rings 25 cycles.

 

I know that I am just a dumb analog hardware engineer, but on our D/A converter we have two filter positions. One is almost a typical "brickwall" (I say "almost" because it is minimum phase instead of linear phase, so there is NO pre-ringing -- which is NEVER found in nature -- only post-ringing which is ALWAYS found in nature) and it has perhaps 10 or 15 cycles of ringing. But when we change to a less steep filter that has only ONE cycle of ringing, the sound quality is MUCH better. Yet you are happy with 25 cycles of ringing.

 

Maybe someday I will understand the advanced level that you are working at. But in the meantime, I remain your truly, just a dumb analog hardware engineer,

Charles Hansen

Dumb Analog Hardware Engineer
Former Transducer Designer

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I've got to say I find Mr. Hansen very convincing and am looking forward to hearing his ADC when it arrives to a nearby location. It's great that a equipment producer such as him takes the time to participate here.

Main listening (small home office):

Main setup: Surge protector +>Isol-8 Mini sub Axis Power Strip/Isolation>QuietPC Low Noise Server>Roon (Audiolense DRC)>Stack Audio Link II>Kii Control>Kii Three (on their own electric circuit) >GIK Room Treatments.

Secondary Path: Server with Audiolense RC>RPi4 or analog>Cayin iDAC6 MKII (tube mode) (XLR)>Kii Three .

Bedroom: SBTouch to Cambridge Soundworks Desktop Setup.
Living Room/Kitchen: Ropieee (RPi3b+ with touchscreen) + Schiit Modi3E to a pair of Morel Hogtalare. 

All absolute statements about audio are false :)

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But by learning the lessons that DSD taught us, we can make high-res PCM with equal or better sound quality than DSD.

 

I think that the most important lesson that DSD taught us, is that going from SDM to any of the commercially available PCM rates results in loss of resolution. I'm actually surprised that all recordings today are not captured directly from the Sigma Delta Modulators (bypassing PCM conversion). I guess that's a lesson some people apparently missed.

 

Tom Caulfield hit the nail on the head when he said that 'the belief that PCM is somehow superior to the original bit stream produced by the Delta-Sigma Modulator in the ADC is the tail wagging the bull.'

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Hi Charles,

 

 

I'm really having fun reading this enthusiast and instructive thread. However, I have the feeling it's getting into a DSD/PCM debate that, though very interesting in itself, is slowly going away from the starting point. Here's my dumb question : I'm using my QB-9 to convert my digital music played through a Mac Book Pro. I've read many times that the USB Vbus power supplied by the computer could be a big problem for sound quality. I think I also read power supply quality is the main reason for which big Mac pros are said to be preferable to Mac Book Pros in a audiophile environment.

What I do for the moment is to simply unplugged my Mac Book from its alimentation in order to make it work with its battery only. I've discovered that it noticeably improves the sound quality.

In the new features you added to the QB-9 you talked of :

 

 

"Addition of an AC line powered supply for the USB circuitry. This provides for uniformly superior performance, regardless of the quality of the USB Vbus power supplied by the computer."

 

 

Does that mean I won't have to unplugged my Mac Book anymore and that its cheap power supply won't be an issue anymore ?

Sorry if my question sounds silly and my english pretty barbaric.

 

 

Thank you again.

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Hello Gus,

 

So the reality of the situation is, whether you believe it or not, there will NEVER be a widespread variety of great musicians on available on DSD.

 

Musically yours,

 

Hello Charles,

 

I sure hope you are wrong about this. I am trying very hard to make exactly that happen.

 

I am sorry if I took your thread off-track, but I had to respond to the Sonoma comments.

 

Your friend,

gus...

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BUT to do so, one should really start with a DSD-128 (double-speed) or DSD-256 (quadruple-speed) recording equipment. And then the engineers must understand the limitations of DSD and be VERY careful how many times they run it through the Sonoma. Standard DSD should not be converted more that ONCE. If they start with special high-speed DSD, they can convert it a few times.

 

Ohhh, so this is why the QA-9 can record in DSD256 while the QB-9 stops at DSD64 (in addition to the near complete lack of titles >DSD64).

 

Thanks again for the highly educational posts.

Roon ROCK (Roon 1.7; NUC7i3) > Ayre QB-9 Twenty > Ayre AX-5 Twenty > Thiel CS2.4SE (crossovers rebuilt with Clarity CSA and Multicap RTX caps, Mills MRA-12 resistors; ERSE and Jantzen coils; Cardas binding posts and hookup wire); Cardas and OEM power cables, interconnects, and speaker cables

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So the reality of the situation is, whether you believe it or not, there will NEVER be a widespread variety of great musicians on available on DSD.

 

Absolute rubbish. Then again, I have no idea of your definition of "great musicians".

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Ohhh, so this is why the QA-9 can record in DSD256 while the QB-9 stops at DSD64

 

No.

 

Originally Posted by Charles Hansen

The reason that we don't do DSD-128 is because it requires that the USB receiver be capable of doing 8 Fs sample rates. We use the XMOS and they make it in two speed grades. Most of our QB-9's have the faster processor, because it is only about $1 more and it is easier to find. But not all of them that faster processor.

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I'm really having fun reading this enthusiast and instructive thread. However, I have the feeling it's getting into a DSD/PCM debate that, though very interesting in itself, is slowly going away from the starting point. Here's my dumb question : I'm using my QB-9 to convert my digital music played through a Mac Book Pro. I've read many times that the USB Vbus power supplied by the computer could be a big problem for sound quality. I think I also read power supply quality is the main reason for which big Mac pros are said to be preferable to Mac Book Pros in a audiophile environment.

What I do for the moment is to simply unplugged my Mac Book from its alimentation in order to make it work with its battery only. I've discovered that it noticeably improves the sound quality.

In the new features you added to the QB-9 you talked of :

 

"Addition of an AC line powered supply for the USB circuitry. This provides for uniformly superior performance, regardless of the quality of the USB Vbus power supplied by the computer."

 

Does that mean I won't have to unplugged my Mac Book anymore and that its cheap power supply won't be an issue anymore?

 

Hello Onkle,

 

You asked and excellent question, and your English is fine. I understand exactly what you are asking, and I am SURE that would not be the case if I tried to answer in French..... :-)

 

We added this power supply because we had a customer who said that when he powered the USB section of the QB-9 with 4 x AA NiMH battery pack (BE CAREFUL IF YOU DO THIS!!! The maximum input to the regulators used is only 5.5 volts. NiMH is only 1.2 volts, so this arrangement supplies 4.8 volts. BUT if you use alkaline cells (1.5 volts each) it will send 6.0 volts and damage the internal voltage regulators!!!). He said that if he used Ever-ready brand the sound was the same, but with DuraCell brand the sound was MUCH better.

 

So we tried the same test, using the exact same battery brands and capacities that he had used. In our system with a MacBook Pro, they BOTH sounded worse than the power from the computer. So we tried to ask him what computer he used but he never responded. The problem is that the only computers we have at work are MacBooks and Lenovo ThinkPads. Both are top quality designs with excellent parts. But we know that many common computers (eg, Dell) do not use such good parts, so we used the AC power. It sounds a TINY bit better than the power from the MacBook Pro, but if you have a computer with a bad power supply, it could sound much better.

 

But your question is somewhat different. When you unplug the AC power supply to the MBP, it is a switching power supply. All switching power supplies generate noise. Now with the updated QB-9-DSD, this noise will not travel through the computer to the USB cable. BUT it will travel into the AC mains in your house. Then it can put noise into the AC power of every component in your house. Some audio components are more sensitive to this noise than others. But even the best will have some effect from the noise on the AC. That is why so many companies make power line filters.

 

However, almost every power line filter that I have ever heard does some things that help the sound and other things that hurt the sound. When you first buy it and install it in your system, you usually only notice the things that help the sound. So the best thing to do is put it in your system and live with it for three weeks. Then when you are used to the way it sounds, take it back out of your system. That is when you will hear all of the ways that it has be damaging the sound. If you have a power filter such as this, that does some good AND some bad, you can either sell it or try putting on your computer equipment or refrigerator or some other place where it won't hurt the sound. These other appliance create a lot of noise and even a bad power line filter can help here.

 

Now I cannot say for certain, because I have not heard every single power line filter in the world. But all of the ones that I have heard, except for one, do some things good and some things bad. You may think I am saying this just to make money, but that is OK. You should always listen for yourself and see what YOU think in YOUR system. Because the only one I have ever heard that only does good things and no bad things is the Ayre power line filter.

 

Please don't make a post that says, "I have an XYZ power line filter and it only does good things!". But IF you have used the Ayre power line filter (especially for at least several weeks -- it takes at least one week just to break in), then feel free to make a post telling me if I am right or if I am wrong. If you have actually used it for a week or more and think it sounds bad, then you have a valid point. But it is one of the "secrets" of the audio world. It cost much less than most filters, so everybody thinks it can't be very good -- but they are wrong. If you have tried it please make a post, either positive or negative.

 

There is one other thing to try. It is a noise filter that plugs into the wall outlet and absorbs the noise from your appliances and computers:

 

Snuffers

 

Unfortunately he only makes them with AC plugs for the US (also Japan and Taiwan use these plugs). You can e-mail him and ask him to make version for Schuko plugs in Europe. He may do it for you -- I don't know. But everything this guy makes (mostly for electric guitars!) sounds excellent.

 

Best regards

Charles Hansen

Dumb Analog Hardware Engineer
Former Transducer Designer

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I think that the most important lesson that DSD taught us, is that going from SDM to any of the commercially available PCM rates results in loss of resolution. I'm actually surprised that all recordings today are not captured directly from the Sigma Delta Modulators (bypassing PCM conversion). I guess that's a lesson some people apparently missed.

 

Tom Caulfield hit the nail on the head when he said that 'the belief that PCM is somehow superior to the original bit stream produced by the Delta-Sigma Modulator in the ADC is the tail wagging the bull.'

 

Hello Hiro,

 

Well, I suppose that if you have equipment that sounds better with large amounts of RFI (all the way to the hundreds of MHz range) injected directly into the input jacks, then you would prefer the straight output of a Delta-Sigma Modulator.

 

However most people go to great lengths to keep RFI out of their audio system. They use things like shielded cables and metal chassis that shield the circuits, and many use power line filters. But I suppose that your system has none of these things because you like the sound of RFI contamination so much....

 

Have fun listening to your RFI,

Charles Hansen

Dumb Analog Hardware Engineer
Former Transducer Designer

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