CHORD ELECTRONICS DAVE
Dec 29, 2016 at 11:24 AM Post #6,406 of 25,873

Can't wait to hear a sample from the Davina,.....hint....hint.....
wink_face.gif

 
Dec 29, 2016 at 1:11 PM Post #6,407 of 25,873
Bit of both - it has two basic modes - recording, so analogue in, digital out. Or post processing, so digital in, digital out but with different sample rates and bit depths. It can decimate or up-sample, or sample rate convert. I will be talking more about it after CES.

The primary intention is to make a 768 kHz 24 bit recording. Then later use Davina to post process this file into any format.

Rob


Interesting Rob!

So it could be a super upsampler for the DAVE with digital in - digital out is what you are saying or? :wink:

Better SQ with DAVINA connected to DAVE
 
Dec 29, 2016 at 1:32 PM Post #6,408 of 25,873
what is Davina??? :O i hope it will be something to be used with Chord DAVE though, extra power hopefully or extra Bass :/ 
 
 
Interesting Rob!

So it could be a super upsampler for the DAVE with digital in - digital out is what you are saying or?
wink.gif


Better SQ with DAVINA connected to DAVE

+1
AMEN!!  
beerchug.gif
 
 
Dec 29, 2016 at 1:46 PM Post #6,409 of 25,873
Bit of both - it has two basic modes - recording, so analogue in, digital out. Or post processing, so digital in, digital out but with different sample rates and bit depths. It can decimate or up-sample, or sample rate convert. I will be talking more about it after CES.

The primary intention is to make a 768 kHz 24 bit recording. Then later use Davina to post process this file into any format.

Rob


Interesting Rob!

So it could be a super upsampler for the DAVE with digital in - digital out is what you are saying or? :wink:

Better SQ with DAVINA connected to DAVE

Seems unlikely: DAVE uses 164,000 tap upsampling to get to 705.6/768 KHz. At best, DAVINA would sound the same, since DAVE can accept 705.6/768 KHz. DAVINA would need to have far more taps in upsampling to better DAVE. Apparently there's no FPGA to do that, with the correct quantity of DSP cores and programmable cells.

So, erm, DAVINA would need to be a multi-FPGA design or be programmed quite differently, to exceed DAVE's primary WTA upsampler.



Now playing: The Philip Glass Ensemble - "Le Domain de la Bete"
 
Dec 29, 2016 at 1:53 PM Post #6,410 of 25,873
  what is Davina??? :O i hope it will be something to be used with Chord DAVE though, extra power hopefully or extra Bass :/ 
 
 
+1
AMEN!!  
beerchug.gif
 

 
The Davina project has been pushing the boundaries of sampling knowledge for many months, and Rob has explained the current status of research, many times. This status is a moving target, so many interesting ideas/opportunities have been explained (searching this thread with the keyword Davina, identifies 160 posts but maybe only a few of them contain the most interesting details). You could find some of them interesting, especially the oft-mentioned hope that recording studios will use Davina, to create music files that will really test the Mojo (and DAVE 
wink.gif
).
 
Dec 29, 2016 at 1:54 PM Post #6,411 of 25,873
My little discovery ... I ask to the expert if they confirms my thesis about correct polarity setting:

Positive polarity (+ correct) or negative polarity (- incorrect).

If the number of tracks on a CD is equal, the first track will be in phase, (+ polarity), the second inverted (- polarity) and so on all alternating: + - + - ...

If the number of tracks on a CD is odd, the first track will be out of phase, (- polarity), the second inverted (+ polarity) and so on all alternating: - + - +...

My conclusion is that this depends on the printing of the CD and its probably the criterion for which the laser of the CD player reads the information about the structure of the CD (number of the tracks...).

 
Say what? Do you have anything of this is an article or something? I'm not saying that your wrong, it's only that I've never heard of such of thing.
 
And the reason that I'm skeptical, without know the technicalities of your assertion, is that one of the biggest examples of -polarity is XTC's Skylarking album, which for years was only available in negative polarity - the whole album! not just even or odd tracks, because it was recorded that way by Todd Rundgren. It finally got a corrected polarity re-release in 2001, but the dynamics were pretty squashed. It was only until this year did Steven Wilson re-mix the album, while also correcting the polarity on the original mix in the form of a flat transfer.
 
This is why your post rings a little false to me - again, not saying that it is - but also, with the polarity function on the DAVE, that means one could not actually get a correct polarity listen of any album, ever, by choosing your own polarity, because polarity alternates between tracks.
 
I would simply like to delve deeper into your post.
 
That said, to my ears, and I know it does not affect the actual SQ, but like the WhatHiFi review, I like negative polarity more than positive, as it seems to make everything fit into a certain pocket. But, I've been listening to positive polarity on most of everything recently, and it sounds damn good, too! So, what do I know? Heh. 
 
Dec 29, 2016 at 2:26 PM Post #6,412 of 25,873
It is mine experience... No any article found...
If you have a very good speakers setup you can here the difference, track by track... As I said, for me, it's not a matter of the recording but the printer info of the CD support... all my CD, all the tracks, sound better if the polarity is correct, and all these track are + - + - or - + - + depending by the total number of the tracks. Never found two track near each other with the same polarity... And always the last track is - (negative).
 
Dec 29, 2016 at 2:48 PM Post #6,413 of 25,873
It is mine experience... No any article found...
If you have a very good speakers setup you can here the difference, track by track... As I said, for me, it's not a matter of the recording but the printer info of the CD support... all my CD, all the tracks, sound better if the polarity is correct, and all these track are + - + - or - + - + depending by the total number of the tracks. Never found two track near each other with the same polarity... And always the last track is - (negative).


Man, from my understanding recordings should just not be this way. Either it was recorded one way or the other. Unless recordings were made between multiple studios, and one or more tracks were not recorded with positive polarity. It would sound to me like it would be an error in CD burning (printing? I take it you are referring to burning audio to a CD-R).
 
Dec 29, 2016 at 2:52 PM Post #6,414 of 25,873
Nope...
All original CD with a perfect Vitus 025 mk2 player...
But if you do not try, you can't believe it...
 
Dec 29, 2016 at 3:16 PM Post #6,415 of 25,873
@Rob Watts

Interesting to hear your version and opinion to this statement from the Ifi Audio team about digital filters and tap lengths, for less reverb ?

Org post:

Originally Posted by Deftone View Post

Switching between mojo and idsd I notice idsd has like a reverb effect, I wonder what it is. No enhancements turned on like bass boost or 3d.

We would not presume to know what you are hearing exactly, but here are some technical basics...

No iFi product includes anything like an intentional reverb effect. The only thing that acts in a matter parallel to a reverb are the digital filters, which are standard on almost all DAC's and which in most iFi products are selectable. The iDSD micro also includes the bit-perfect mode which shuts down digital filters.


TL;DR

Digital filters trade off flat frequency response to the nyquist frequency of the recording for transient distortion. The "longer" a filter (the more so-called taps it has) is the greater the impulse/transient distortion. In the iDSD micro the standard filter is the "longest" whereas the bit-perfect mode is the "shortest".

Audibility of different digital filters is a very steamy topic, some people are very sensitive to these and others are not. For those in the former group, we've included the option to select a filter that suits their personal taste and preference.


The long version

The "standard" digital filter is a sharp roll-off FIR type. This one is made by creating a delay chain with "taps" at each 1 sample delay and then applying different "gain" to the signal at each tap. The result of each tap is summed together to create a new signal that has "unwanted" ultrasonic content removed. More on FIR digital filters may be read here:

https://en.wikipedia.org/wiki/Finite_impulse_response

The upshot is that a digital filter (be it FIR or IIR) actually introduces a form of reverb. The more taps a digital filter has, the longer the reverb. The "standard" filter in the iDSD micro has around 256 taps (which is industry standard) and it's complete "reverb window" covers a time window of around 0.7 milliseconds at 44.1kHz source material.

Human hearing acuity allows the detection of transient acoustic events of 0.01 milliseconds (10 uS) in length while the detection of a tone burst will require around 2 milliseconds at least.

This means that the standard digital filter is likely to be at the edge of audibility for transient events (e.g. a snare drum rim shot) but should be inaudible on tone bursts (e.g. a piano note). More on the temporal resolution of the human hearing is to be found here:

http://biology.stackexchange.com/questions/27662/what-is-the-human-ears-temporal-resolution

Some people advocate digital filters with very large numbers of taps. For arguments sake, if we'd have used 16384 taps at 44.1kHz, the "reverb window" would cover 46.5 milliseconds, which one would expect to have a significantly audible effect on both transients and tone bursts.

Using the "minimum" filter (actually a Bezier type) with only around 64 taps, it's complete "reverb window" covers a time window of around 0.17 milliseconds at 44.1kHz source material. This may still be audible on transients, but should be reliably inaudible on tone bursts.

Finally the bit-perfect filter has just 1 tap, which translates to no delay chain and no "reverb". But the time domain resolution of 44.1kHz PCM recordings limit the impulse response to 0.0227 milliseconds. Said number is still not in a class that could be considered as reliably inaudible, which is one of the reasons why recordings with a sample rate higher than 44.1kHz are desirable.

What the filter switch on the iDSD micro (and on other iFi products) offers to the customer is the ability to select her/his personal preference of the trade-offs involved.

#7239 of 7241
an hour ago
iFi audio
Quote:
Originally Posted by Deftone View Post


I understand what your saying but this isn't subtle reverb like I hear on mojo and other quality DACs, this is very pronounced and causes even sound effects in Windows on notifications for example to echo when it's not supposed to or on "fake" electronic samples of music that doesn't have it naturally.

We advise to check your Windows audio device settings and make sure that "disable all enhancements" checkbox is selected. Normally this OS should not enable any of these by itself after installing a new sound device, but it can be funny at times.

Also please check any enhancement/effect settings in your playback software, in case these got accidentally turned on.
 
Dec 29, 2016 at 9:09 PM Post #6,416 of 25,873
@Rob Watts

Interesting to hear your version and opinion to this statement from the Ifi Audio team about digital filters and tap lengths, for less reverb ?

Org post:

Originally Posted by Deftone View Post

Switching between mojo and idsd I notice idsd has like a reverb effect, I wonder what it is. No enhancements turned on like bass boost or 3d.

We would not presume to know what you are hearing exactly, but here are some technical basics...

No iFi product includes anything like an intentional reverb effect. The only thing that acts in a matter parallel to a reverb are the digital filters, which are standard on almost all DAC's and which in most iFi products are selectable. The iDSD micro also includes the bit-perfect mode which shuts down digital filters.


TL;DR

Digital filters trade off flat frequency response to the nyquist frequency of the recording for transient distortion. The "longer" a filter (the more so-called taps it has) is the greater the impulse/transient distortion. In the iDSD micro the standard filter is the "longest" whereas the bit-perfect mode is the "shortest".

Audibility of different digital filters is a very steamy topic, some people are very sensitive to these and others are not. For those in the former group, we've included the option to select a filter that suits their personal taste and preference.


The long version

The "standard" digital filter is a sharp roll-off FIR type. This one is made by creating a delay chain with "taps" at each 1 sample delay and then applying different "gain" to the signal at each tap. The result of each tap is summed together to create a new signal that has "unwanted" ultrasonic content removed. More on FIR digital filters may be read here:

https://en.wikipedia.org/wiki/Finite_impulse_response

The upshot is that a digital filter (be it FIR or IIR) actually introduces a form of reverb. The more taps a digital filter has, the longer the reverb. The "standard" filter in the iDSD micro has around 256 taps (which is industry standard) and it's complete "reverb window" covers a time window of around 0.7 milliseconds at 44.1kHz source material.

Human hearing acuity allows the detection of transient acoustic events of 0.01 milliseconds (10 uS) in length while the detection of a tone burst will require around 2 milliseconds at least.

This means that the standard digital filter is likely to be at the edge of audibility for transient events (e.g. a snare drum rim shot) but should be inaudible on tone bursts (e.g. a piano note). More on the temporal resolution of the human hearing is to be found here:

http://biology.stackexchange.com/questions/27662/what-is-the-human-ears-temporal-resolution

Some people advocate digital filters with very large numbers of taps. For arguments sake, if we'd have used 16384 taps at 44.1kHz, the "reverb window" would cover 46.5 milliseconds, which one would expect to have a significantly audible effect on both transients and tone bursts.

Using the "minimum" filter (actually a Bezier type) with only around 64 taps, it's complete "reverb window" covers a time window of around 0.17 milliseconds at 44.1kHz source material. This may still be audible on transients, but should be reliably inaudible on tone bursts.

Finally the bit-perfect filter has just 1 tap, which translates to no delay chain and no "reverb". But the time domain resolution of 44.1kHz PCM recordings limit the impulse response to 0.0227 milliseconds. Said number is still not in a class that could be considered as reliably inaudible, which is one of the reasons why recordings with a sample rate higher than 44.1kHz are desirable.

What the filter switch on the iDSD micro (and on other iFi products) offers to the customer is the ability to select her/his personal preference of the trade-offs involved.

#7239 of 7241
an hour ago
iFi audio
Quote:
Originally Posted by Deftone View Post


I understand what your saying but this isn't subtle reverb like I hear on mojo and other quality DACs, this is very pronounced and causes even sound effects in Windows on notifications for example to echo when it's not supposed to or on "fake" electronic samples of music that doesn't have it naturally.

We advise to check your Windows audio device settings and make sure that "disable all enhancements" checkbox is selected. Normally this OS should not enable any of these by itself after installing a new sound device, but it can be funny at times.

Also please check any enhancement/effect settings in your playback software, in case these got accidentally turned on.

 
The taps length and # of taps are dependent on the digital filter implementation. Sounds like iFi is generalizing that the digital filter taps and tap lengths are a function of "reverb". I bet Rob's WTA filter is completely different from iFi's and more taps and tap lengths are desirable to reconstruct the analog signal WITHOUT adding fake reverb, fake details etc. IMO, the strongest points in Rob's DACs are incredible timing that does not miss a beat without adding fake details, reverb, etc. while other delta sigma DACs with their inferior digital filter implementation just adds fake details and reverb. NOS R2R DACs OTOH do not even use a digital filter thus the timing and reconstruction of analog signal must be ultra precise in order to not cause distortion/fake details/reverb etc.
 
All of the statements above are of course my opinion on the subject.
 
Dec 29, 2016 at 9:15 PM Post #6,417 of 25,873
Interesing that on the iFi iDSD thread, someone just confirmed my statement regarding iFi's digital filter producing lots of reverb while Rob's WTA filter OTOH are have amazing timing and a more realistic timbre:
 
 
I don't have micro but I have nano and mojo and experienced same thing, I noticed this straight away after upgrading to mojo, my explanation is that nano has very wet/lean sound and mojo has dry/fast, maybe too dry sound and too fast sound (the sounds stops very quickly, quicker than your mind can think of). You can think as micro having longer decays and mojo having none (which is true in my perception). The decays are filled with DETAILS.
Imagine a 1 second sound split in 10 equal units. Nano has sound between 0.0 and 0.1, between 0.2 and 0.3, between 0.4 and 0.5, between 0.6 and 0.7, between 0.8 and 0.9. Mojo fills the gaps with more details and nano fills them with decays.
Many said that micro has wet sound so it's not something new.
In my perception, Micro was created for fun pleasure, it was not meant for accuracy or realism. That's why it has so many filters and bass boost and 3d and such things should not exist in a DAC. It's like using different EQ on different songs. "I don't like this song with bass boost, I don't want this song with 3D", if they included such things, it should work with every song.
Micro makes sense in a stereo hi-fi setup, where you need to hear better the emotion in voice of the artist. Most Hi-fi speakers have built-in wet sound so having a wet DAC, can make them too lean, but for slow music like jazz and classical, I guess this is favorable. Hi-fi gear is for pleasure, that's what I'm saying. The opposite are the studio monitors which offers dry sound.


After more listening I come to the conclusion

iFi idsd black - brighter, slower, lots of reverb, detailed. Better for very slow female vocal music, jazz etc (not for me)

Chord mojo - flat, lightning fast, more detail, realism and timbre. (Definitely for me) I don't like the typical "audiophile" music I love hard hitting fast metal.

Both units are very good
smily_headphones1.gif

 
http://www.head-fi.org/t/728236/ifi-idsd-micro-dsd512-pcm768-dac-and-headphone-amp-impressions-reviews-and-comments/7230#post_13126693
 
Dec 29, 2016 at 9:25 PM Post #6,418 of 25,873
I remember Rob's statement where in he said impulse response measurement is an illegal signal which does not exist in reality in music. more taps means longer the post and pre echo shown by impulse response ( chord uses symmetrical with equal post and pre echo) so longer post and pre echo in case of large tap lengths shown by impulse response does not matter because such type of response does not exist in reality but the extra tap length helps in constructing the the timing of transients more accurately. infinite tap length would perfectly recreate the analog wave but with infinite post and pre echo shown by impulse response.
 
Dec 30, 2016 at 1:34 AM Post #6,420 of 25,873
 
I am not expert so a question from rob. can Davina record directly from mic will it need an external preamp ( or internal preamp in Davina ) ?


Hope you don't mind me jumping in here. Obviously a dynamic mic (like Sure SM58) doesn't need phantom power but with a condenser mic, for instance 'Neumann' it is necessary. You probably knew that but producers/engineers are pretty particular as to what mic pre's they prefer and so unless Chord are going to license something like Focusrite grade A pre's, adding your own pre's and targeting top end studios would probably result in a pretty superfluous add-on imo.

Its a configurable input - you can set it up as mic input with switchable 48v phantom, or set to line level input.
 
I did not want an extra stage (mic pre-amp) degrading transparency, so hence the configurable nature of the input.
 
Rob 
 

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