[Christer]

I don't know, that Bombay Sapphire headphone stand is pretty inventive

Well regarding the Bombay Sapphire bottle I have to admit it is not my own invention.
I actually saw it somewhere here at Headfi long ago and it made sense to me and works well,so when this bottle was empty I immediately saw a good use for it.

Why waste money on expensive tropical hardwood headphone stands?

I just noted that Grado have a new headphone with earcups made with the South American tropical hardwood Cocobolo.
I would hesitate more than twice before buying those irrespective of SQ.

And those specially designed and expensive Chord stands although metal, are not for me either.
I only buy stuff that makes a difference and improvement SQ wise.

I primarly listen to my equipment. Fancy looks is for the vanity crowd imho.

Which brings me back to the Benchmark dac/amp, which doesn't look like much compared to some "Goldy Looks" big beefy headphone amps that cost three- four or ten times as much but too often ,do not sound any better or as transparent as the Benchmark amp does.
Apart from their new bigger HPA4 few headphone amps few others measure even nearly as well as the HPA2.
There is a good reason some Pro's in the classical recording business use Benchmark amps.

Cables often make a very easy to detect and clearly audible difference to me.
Hence my interest in the Wave BNC cables.
Cheers CC

 

[Christer]

Is there a hifi equivalent to Feng Shui? If there is, that ain’t it!

Well, personally I normally withhold my opinions until I have actually auditioned various equipment, dacs ,cables headphones amps and such.

Have you also done that?

I did not comment at all on any benefits or lack of such with these cables until I had actually auditioned them.

And if my memory serves me right, it normally does, even the designer behind DAVE, RW personally recommended trying ferrites on the BNC cables to combat RF problems when people here began hearing such problems with DAVE/BLU2.
I auditioned DAVE/BLU2 in Singapore both with and without ferrites on the BNC cables and the difference was clearly obvious.

Judging from what I can personally now hear with HMS/Qutest and ferrited insulated cabling it may not only be a simple case of Feng Shui as you seem to think?

I can firmly say that to me, the stock cable is NOT as good as either of these two and possibly several other BNC cables mentioned on this thread lately.

Maybe the Opto DX is even better?
But I won't know until I can personally compare.

Cheers CC

 

[George Hincapie]

Well regarding the Bombay Sapphire bottle I have to admit it is not my own invention.
I actually saw it somewhere here at Headfi long ago and it made sense to me and works well,so when this bottle was empty I immediately saw a good use for it.

Why waste money on expensive tropical hardwood headphone stands?

I just noted that Grado have a new headphone with earcups made with the South American tropical hardwood Cocobolo.
I would hesitate more than twice before buying those irrespective of SQ.

And those specially designed and expensive Chord stands although metal, are not for me either.
I only buy stuff that makes a difference and improvement SQ wise.

I primarly listen to my equipment. Fancy looks is for the vanity crowd imho.

Which brings me back to the Benchmark dac/amp, which doesn't look like much compared to some "Goldy Looks" big beefy headphone amps that cost three- four or ten times as much but too often ,do not sound any better or as transparent as the Benchmark amp does.
Apart from their new bigger HPA4 few headphone amps few others measure even nearly as well as the HPA2.
There is a good reason some Pro's in the classical recording business use Benchmark amps.

Cables often make a very easy to detect and clearly audible difference to me.
Hence my interest in the Wave BNC cables.
Cheers CC

I've never listened to any Benchmark HPAs. You're a fan then?

Cable wise; I'm not convinced. Have spent £1000's on Reference ICs, but now using Blue Jeans cables after that YT video of the guy proving cables make no difference as long as they are well made. Can't say I can tell a difference. But that's me - not projecting on to anyone else

 

[George Hincapie]

Well, personally I normally withhold my opinions until I have actually auditioned various equipment, dacs ,cables headphones amps and such.

Have you also done that?

I did not comment at all on any benefits or lack of such with these cables until I had actually auditioned them.

And if my memory serves me right, it normally does, even the designer behind DAVE, RW personally recommended trying ferrites on the BNC cables to combat RF problems when people here began hearing such problems with DAVE/BLU2.
I auditioned DAVE/BLU2 in Singapore both with and without ferrites on the BNC cables and the difference was clearly obvious.

Judging from what I can personally now hear with HMS/Qutest and ferrited insulated cabling it may not only be a simple case of Feng Shui as you seem to think?

I can firmly say that to me, the stock cable is NOT as good as either of these two and possibly several other BNC cables mentioned on this thread lately.

Maybe the Opto DX is even better?
But I won't know until I can personally compare.

Cheers CC

Why not just use AES and not worry about EMI at all?

 

[naynay]

For me currently, the question is whether to Stream or to Storm

Are you saying you can't hear a £515 difference?

 

[George Hincapie]

Are you saying you can't hear a £515 difference?

Nice system.

 

[Christer]

I've never listened to any Benchmark HPAs. You're a fan then?

Cable wise; I'm not convinced. Have spent £1000's on Reference ICs, but now using Blue Jeans cables after that YT video of the guy proving cables make no difference as long as they are well made. Can't say I can tell a difference. But that's me - not projecting on to anyone else

Hello , no I'm not a fan of ANY HIFI brand.
Or any other brand for that matter.
But my connections in the pro field is sometimes more useful to me, than advice from audiophiles.
I listen and compare and choose the equipment I can afford or I am willing to pay for ,and which sounds realistic with real live acoustic music to me.
Cables have to work with other things in a system and can make quite a big difference imho.
I have no technical explanation at all with regard to rca or now BNC cables but the different combinations of rca and BNC I can now connect all sound different.
Sometimes a big difference .Sometimes a smaller one, but still an audible one with my reference tracks where I was actually at the sessions and in both the hall and the control room.
Cheers CC

 

[Christer]

Are you saying you can't hear a £515 difference?

Well if all my recordings were two three mics perfectly placed in a very good hall,coherent recordings it would not be so tough to decide.But they are NOT so I am sort of going through a numbers game.
Cheers CC

 

[Christer]

Why not just use AES and not worry about EMI at all?

Hmm. AES as in...?
Cheers CC

 

[George Hincapie]

AES ICs. Balanced. Design makes them immune to EMI AFAIK. Cue someone telling me I am wrong in 3, 2, 1

 

[audio_1]

AES ICs. Balanced. Design makes them immune to EMI AFAIK. Cue someone telling me I am wrong in 3, 2, 1

Single ended components sound better in a domestic installation with short interconnects where noise is not an issue imho. Half the number of components in the signal path. All Chord dacs are single ended. Chord also recommend single ended digital connections. This is why Chord dacs have dual BNC connections instead of dual AES.

 

[me2621a]

@Rob Watts

I recently watched the talk you gave where you explained the M Scalers technology and the benefits of reaching a million taps with the WTA algorithm. I had three questions I was hoping you could answer:

1. If I understood your talk correctly, the benefit of the M Scaler and the ability for it to run the WTA filter with over a million taps is that you are able to reconstruct the original analog sine wave to a 16 bit depth from a 44/16 source. If I understood that correctly, what I do not understand is why that helps the downstream DAC, as the M Scaler is not responsible for the actual conversion of digital to analog, and the Dave for example only has ~160k taps, and therefore cannot reconstruct the sine wave during the actual D -> A conversion (it can approximate it with great accuracy as we all know the Dave can of course but it could always do this). Which leads me to my questions which is: why does the M Scaler make the downstream DAC sound better (to be clear I am not debating if the M Scaler makes the downstream Dac sound better, I have heard it, it does I just don’t quite follow why)? Is it because the M Scaler is able to reconstruct the original wave, and resample the original wave at 700+KHZ and therefore the downstream DAC has much more information to work off of or is there something that I am not understanding?

2. At 700+ KHZ are timing errors now bellow the brains ability to notice errors with transients?

3. When running higher resolution files through the M Scaler (eg: 96/24), does the M Scaler have to do less or more work? Asked a different way (assuming my understanding in 1 is correct), because the M Scaler can reconstruct the original sine wave with a 44/16 source, does that mean it can do so with a 96/16 source (ie 96/16 is easier because there is more data to reconstruct from), or can it not? Likewise for 24 bit data, is the M Scaler just not able to mimic the sinc function (I believe that is what you said the WTA algorithm approximates down to 16 bits) to 24 bits, and therefore would require more taps to do so?

Thank you for taking the time to read/answer.

 

[Rob Watts]

@Rob Watts

I recently watched the talk you gave where you explained the M Scalers technology and the benefits of reaching a million taps with the WTA algorithm. I had three questions I was hoping you could answer:

1. If I understood your talk correctly, the benefit of the M Scaler and the ability for it to run the WTA filter with over a million taps is that you are able to reconstruct the original analog sine wave to a 16 bit depth from a 44/16 source. If I understood that correctly, what I do not understand is why that helps the downstream DAC, as the M Scaler is not responsible for the actual conversion of digital to analog, and the Dave for example only has ~160k taps, and therefore cannot reconstruct the sine wave during the actual D -> A conversion (it can approximate it with great accuracy as we all know the Dave can of course but it could always do this). Which leads me to my questions which is: why does the M Scaler make the downstream DAC sound better (to be clear I am not debating if the M Scaler makes the downstream Dac sound better, I have heard it, it does I just don’t quite follow why)? Is it because the M Scaler is able to reconstruct the original wave, and resample the original wave at 700+KHZ and therefore the downstream DAC has much more information to work off of or is there something that I am not understanding?

2. At 700+ KHZ are timing errors now bellow the brains ability to notice errors with transients?

3. When running higher resolution files through the M Scaler (eg: 96/24), does the M Scaler have to do less or more work? Asked a different way (assuming my understanding in 1 is correct), because the M Scaler can reconstruct the original sine wave with a 44/16 source, does that mean it can do so with a 96/16 source (ie 96/16 is easier because there is more data to reconstruct from), or can it not? Likewise for 24 bit data, is the M Scaler just not able to mimic the sinc function (I believe that is what you said the WTA algorithm approximates down to 16 bits) to 24 bits, and therefore would require more taps to do so?

Thank you for taking the time to read/answer.

Good questions, and not so easy to answer.

1. Take a regular 48k/24b recording; that has samples every 20.8 uS. We know that the ear/brain is at least as sensitive to 4 uS (my subjective evidence suggests it's actually a great deal more sensitive). Also, we know that if we use a sinc function we will perfectly reconstruct the original bandwidth limited signal; we also know that if we do not use a sinc function the transients will not be reconstructed accurately - they will be a little too early or too late. We also know from psychoacoustics that transients are essential from a perception POV, as they are used by the brain to compute pitch, timbre, rhythm and instrument separation. The M scaler takes the 48k signal and then converts it too 768k with a guaranteed 16 bit accuracy - as the M scaler is the same as an ideal sinc function to better than 16 bits - so it will recover the bandwidth limited analogue signal to better than 16 bits. We now have a 768k recording, with an output of every 1.3 uS now, which of course is much easier for the brain to deal with.

2. But some questions remain. Every time we double the sample rate, the size of the area of the transient error gets 4 times smaller; and with a 16 times increase in sample rate we have a 256 times reduction in the area of the error - but even with a 768 k output we can still get a small transient error. Now this is still audible, and that's where the second WTA filter in my DACs takes over - and this takes the 768k signal and converts it too 12,288 kHz, or a sample every 88 nS, and this filter is still audible, even though we are now dealing with very small timing errors. So yes timing errors above 768k are subjectively important - and it's an aspect I am still researching into. We will eventually hit two barriers - there is no point in getting more accurate transients if the DAC can't accurately resolve it when the DAC converts it too analogue - and there is no point in reconstructing transients to a higher accuracy than the ear/brain can detect. It's these aspects I am looking into.

But there is something special about hitting 768k. The 768k recordings I have sound very different to 384k recordings, and have the qualities that the M scaler gives; this suggests that there is something special about hitting the 1.3us accuracy of 768k. And certainly, the WTA 2 filter currently does not sound like the M scaler - it only seems to affect the ability to perceive the starting and stopping of notes, not pitch, timbre and instrument separation, which happens when you get to 768k.

3. From a processing POV going from 384 to 768 involves exactly the same number of multiplications and processing as 48k to 768k; it's just the filter updates its input samples at a much faster rate, but the M scaler does exactly the same work. Also, the benefit of 16 bit reconstruction when using 16 bit files is obvious - but with 24 bits what do we need? Again, this is another aspect that I am researching into.

I hope that clarifies it for you. In short there are indeed more questions that need to be answered...

 

[ZappaMan]

And certainly, the WTA 2 filter currently does not sound like the M scaler - it only seems to affect the ability to perceive the starting and stopping of notes, not pitch, timbre and instrument separation, which happens when you get to 768k.

Could you elaborate a little why you are comparing wta2 to mscaler?
And is that wtA from a certain device or are all implementations of wta2 the same?

 

[Christer]

Good questions, and not so easy to answer.

1. Take a regular 48k/24b recording; that has samples every 20.8 uS. We know that the ear/brain is at least as sensitive to 4 uS (my subjective evidence suggests it's actually a great deal more sensitive). Also, we know that if we use a sinc function we will perfectly reconstruct the original bandwidth limited signal; we also know that if we do not use a sinc function the transients will not be reconstructed accurately - they will be a little too early or too late. We also know from psychoacoustics that transients are essential from a perception POV, as they are used by the brain to compute pitch, timbre, rhythm and instrument separation. The M scaler takes the 48k signal and then converts it too 768k with a guaranteed 16 bit accuracy - as the M scaler is the same as an ideal sinc function to better than 16 bits - so it will recover the bandwidth limited analogue signal to better than 16 bits. We now have a 768k recording, with an output of every 1.3 uS now, which of course is much easier for the brain to deal with.

2. But some questions remain. Every time we double the sample rate, the size of the area of the transient error gets 4 times smaller; and with a 16 times increase in sample rate we have a 256 times reduction in the area of the error - but even with a 768 k output we can still get a small transient error. Now this is still audible, and that's where the second WTA filter in my DACs takes over - and this takes the 768k signal and converts it too 12,288 kHz, or a sample every 88 nS, and this filter is still audible, even though we are now dealing with very small timing errors. So yes timing errors above 768k are subjectively important - and it's an aspect I am still researching into. We will eventually hit two barriers - there is no point in getting more accurate transients if the DAC can't accurately resolve it when the DAC converts it too analogue - and there is no point in reconstructing transients to a higher accuracy than the ear/brain can detect. It's these aspects I am looking into.

But there is something special about hitting 768k. The 768k recordings I have sound very different to 384k recordings, and have the qualities that the M scaler gives; this suggests that there is something special about hitting the 1.3us accuracy of 768k. And certainly, the WTA 2 filter currently does not sound like the M scaler - it only seems to affect the ability to perceive the starting and stopping of notes, not pitch, timbre and instrument separation, which happens when you get to 768k.

3. From a processing POV going from 384 to 768 involves exactly the same number of multiplications and processing as 48k to 768k; it's just the filter updates its input samples at a much faster rate, but the M scaler does exactly the same work. Also, the benefit of 16 bit reconstruction when using 16 bit files is obvious - but with 24 bits what do we need? Again, this is another aspect that I am researching into.

I hope that clarifies it for you. In short there are indeed more questions that need to be answered...

Thanks very interesting indeed to read.
My VERY limited knowledge of digital theory does not allow me to follow and understand everything in absolute detail. But I DO know what can hear via my HMS/Qutest and it has fundamentally changed my take on rbcd 16/44.1.

This morning I have just listened to an early digital BIS recording from Gothenburg with the GSO playing Sibelius's 7th symphony with great pleasure!
The recording was made with only THREE Neuman mics in the good acoustics of the Gothenburg hall with SONY's PCMF1 in 1984!
And what I have just heard sounded almost like hearing a mic-feed from that hall.
Playback via PCM F1 did not sound that good in those days.
Strings could sound a bit steely via that ADC.

I am amazed on a daily basis at how many more recordings I can now really enjoy not only for their musical and performance qualities, but also how very realistic 16/44.1 can sound via HMS.

I mentioned those early digital recordings in some posts here after my return from Asia earlier, and via the stock BNC they did NOT sound as good as I had hoped. But now via upgrade BNC cables they do.
And the difference is not subtle to me.

The coherence and realism,how this orchestra actually sounded playing live in their good hall with these early three mics only 16/44.1 recordings via HMS actually sound more realistic to me than BIS's multimic'd Sibelius symphonies recent remakes from Minnesota.
I was invited to those sessions too, but I would have had to pay for my own flight so I declined the offer.
But I have been to quite a few recordings sessions and many concerts in Gothenburg.
It truly sounds as if I am listening to the analogue signal before digitizing with many of those early cd's now.

But yes really well recorded hi res can sound even more realistic, which brings me to the question of which ADC can record at 768khz?
Didn't you mention 256M taps needed to recover 24 bits to the same level as 1M taps can with 16/44.1 two years ago?

The only PRO ADC's which I am familar with are still limited to 24/358 or DSD 256 which I have heard in use at recording sessions.
Have you already made test recordings with a prototype Davina or from which ADC did you get native 768 recordings?
Cheers CC