Oct 16, 2016 at 6:30 AM Post #5,146 of 27,054
This article summarises what's required to make headphones "do 3D":

http://www.innerfidelity.com/content/aes-headphone-technology-conference-highlight-paper#pJsiRF6bqBqihwkF.97

His conclusion is that the only way to make very accurate studio monitor headphones is to first tune the headphone to the diffuse field response—as it delivers the least linear distortion in the transducer/ear interface and will be able to most accurately play an incoming signal for the ear. Then, using digital signal processing (DSP), create a fake room using binaural room impulse response information for a high acoustic quality listening room. Then create virtual speakers in that room to play the sound. Then add a head tracker and a bunch of HRTF data so that you can move your head normally and hear the cues change—because your brain won't be reliably fooled if you don't. Research shows that if you do all these things, only then can you properly perceive tonal neutrality on headphones.


This is basically what the Smyth Realiser A16 does:

http://www.head-fi.org/t/807459/smyth-research-realiser-a16

and it has no difficulty reproducing the soundstage that a Dolby Atmos speaker setup can portray. Essentially it's possible for headphones, with a correctly derived personalised head related transfer function to completely disappear and render a fully realistic sound field.

I've listened to the A16. It's extraordinary.
 
Oct 16, 2016 at 6:31 AM Post #5,147 of 27,054
@x RELIC x , yes exactly. binaural recordings already incorporate the delay perceived by the two ears and headphones allow that recording to be heard by each ear without adding further delay or any further interaction because headphones place the transducers closest possible to ears unlike speakers but then binaural recordings have its own limitations due to physics.
 
Oct 16, 2016 at 6:35 AM Post #5,148 of 27,054
Sound quality does not matter.
 
By this I mean it is not the most important thing - its the ability to get emotional satisfaction from music (musicality) that is primary.
 
And I get to "enjoy" lots of über DAC's at shows - and for me - and of course this is highly personal - they fail big time as regards musicality. I get more musical satisfaction from little old Mojo than any of these über DAC's do.
 
Moreover, I am absolutely convinced that Mojo (ignore Dave for the time being) will beat any other DAC at any price point when it comes down to enjoying music. Why am I so convinced? Because there are sound technical and objective reasons why this would be the case. Let us consider the three most important things that is important for a DAC/amplifier:
 
 
1. Timing - if you spend any time researching psycho-acoustics you will appreciate that timing is the most important parameter that the brain uses to process the data from the ears. It is used for location, timbre, and being able to perceive the starting and stopping of notes. So when I talk about timing problems, what do I mean? Now I am NOT talking about phase errors, or timing differences with frequency response - these are linear errors, and are inconsequential - the brain is used to dealing with these kinds of problems. What I am talking about is when the timing of transients varies either with amplitude (a transient will have a different time depending on whether it is small amplitude or a large amplitude signal) or when the timing of a transient depends upon when it is sampled by the ADC. So if a transient crosses through zero half way between samples, differing interpolation filters will not recreate the timing of the transient accurately, and this sampling dependent uncertainty is highly audible. So how big is the problem of timing? Looking at all the many listening tests I have done, my contention is that a DAC/amp must be accurate down to tens of nS in that surprisingly small timing errors are very audible. 
So what DAC's have minimal timing problems? Let's look at the interpolation filter first, something that is in all DAC's (yes even NOS DAC's have an interpolation filter). Now the job of the interpolation filter is to reconstruct the timing of transients and to remove the HF images of the signal that extends to infinite frequencies that is due to the sampling process. In short, it is part of the process that takes the discontinuous digital data and converts it back into a continuous analogue signal. The better the interpolation filter, the more accurate this is done (closer to the original analogue signal in the ADC before it was sampled). Now the accuracy of the filter in terms of how well transient timing is reconstructed depends upon the filter tap length, and the type of filter. NOS filters are by far the worst, with timing errors of up to a hundred uS. A NOS filter (this is a bad term, all DAC's oversample) is actually a zero order hold interpolation filter, where the oversample rate is determined by the DAC that is used. Because the timing errors are so huge, they sound very soft (when you get timing errors, the brain can't deal with the data, so it can't actually perceive the starting and stopping of notes - and if you can't hear the starting of a note, it sounds soft or out of focus). Now some people like this; but it is clearly artificial and is categorically not transparent. The usual FIR filter is a half band interpolation filter, with a tens of taps. This filter is better than zero order hold (NOS) for timing, but still has massive errors, as there is considerable measurable sampling image errors. These filters return the original sample unchanged, and they are cheap to implement which is why they are used for 99% of the time. Apodizing filters can offer better performance, but they still have substantial timing errors.
So how can you reconstruct the timing of transients perfectly? If you look at sampling theory, it has been proven that a Whittaker-Shannon interpolation filter will return a bandwidth limited sampled signal absolutely perfectly; there will be no timing or amplitude errors at all. But to have an ideal interpolation filter you need a sinc impulse response and many many taps; to absolutely guarantee just 16 bit performance under all circumstances the coefficients need to be smaller than 16 bits - and this happens at about 1 million taps for a 16 FS filter. And every listening test I have ever done gives the same conclusion - more taps, better and more transparent sound quality. Even with Dave at 164,000 taps, we have not reached the subjective limit yet.
So onto the next timing problem - amplitude errors. Now R2R DAC's have slow FET's to switch the resistors in and out, and these themselves cause their own glitch issues. Moreover, they are very slow - you can only go to 16FS max - so that means the timing is limited to only 1.3 uS. DSD dac's have very large amplitude timing errors too - a small signal has a much bigger delay than a large signal. Currently, the only technique that has the smallest amplitude related timing error is pulse array, as it runs 5 bits at 104 MHz - many times faster than any other DAC topology.
So to solve the timing issues you absolutely need extremely fast DAC's, and very large tap lengths. Even Mojo easily beats all other non Chord DAC's in this regard.   
 
2. Small signal resolution. Clearly small signals are vital - if you can't hear the tiny details, then it no longer becomes a believable performance. Small signals also are used for depth perception, something that I am personally very interested in. Go to a cathedral and listen to an organ at 100 m away; it sounds 100 m away - but reproduced audio is severely depth truncated. Now small signal linearity is measured using fundamental linearity tests - you measure the amplitude of a -60db, set that value as a reference, then measure at -120 dB say. It should be exactly -120.000 dB, but a real DAC won't be. Delta sigma or DSD will actually attenuate the level, R2R will have random errors due to resistor tolerance problems. When it comes to depth perception, there is something extremely strange - any small signal amplitude error (no matter how small) affects depth perception. That's why Dave has 350 dB performance noise shaping, as increasing the performance of the noise shaper gave much better depth performance. Now R2R DAC's have easily measurable small signal errors; DSD is only -120dB accurate noise shaping; conventional DAC noise shapers are about -140dB; Mojo is at -200 dB, and Dave takes the record at -350 dB. I have published FFT's showing Dave's noise shaper performance.
Now this issue is more complex than this; you also need simple analogue stages to further improve performance, you additionally you need a distortion performance that gets better as the signal get smaller.
 
3. Noise floor modulation. This is another can of worms, as DAC's have a multitude of problems from different areas for this. Noise floor modulation is subjectively very important; it is when the noise floor changes with signal level. All conventional DAC's have large amounts of noise floor modulation, and it is my contention that it is very audible - it adds a hardness or grain to the sound. Even the tiniest amount affects smoothness and refinement. With Mojo, we have no measurable noise floor modulation - and similarly Dave but with an even lower noise floor. I have not seen any other DAC come close to this performance.
 
This gives you a brief flavour of the issues involved in producing a truly transparent DAC; and for me it's only by having a truly transparent DAC that one can get musicality. But of course some people have very different tastes, and respond differently to music. Some people like a particular sound; I have done some listening sessions when I drew the complete opposite opinion to somebody else. Some people like distortion; others like an overly soft warm sound. Whatever floats your boat I guess.
 
Rob
 
Oct 16, 2016 at 6:41 AM Post #5,149 of 27,054
Jawed, have you compared the Smyth Realizer A16 to Darin Fong's Out of Your Head software?
 
Oct 16, 2016 at 7:20 AM Post #5,150 of 27,054
I just realized I have said a few things about Chord DACs in private locally that I have never posted here. I am biased and I love the latest generation of Chord DACs from Mojo to DAVE. I explain to local audiophile friends that if we understand the design of Chord DACs, we can ask ourselves following question: Rob Watts could have designed any DAC but he chose to design Chord DACs the way it is done. Why?
 
Rob is a master of programming FPGAs for DACs. If he thinks DSD/PWM is the best way to go, he could have just programmed an FPGA to convert to 2xDSD, 10xDSD or even 32xDSD. He would not need to design a 20-element Pulse Array DAC to follow it. In fact, he probably could have done it 10 years ago if he wanted to and that design would probably be superior to some of the current DSD DAC designs. If he thinks R2R is the best way to go, once again, he could have just programmed an FPGA to upsample everything to 16fs and feed it to an R2R DAC and then work on everything else that goes with an R2R DAC. Or if he thinks DAC chips are good enough, he could have just programmed an FPGA to feed a DAC chip at the sample rate of his choice and then optimize the performance he can squeeze from the DAC chip.
 
I think in general, people go out of their way to design something complex because they believe there is something to be gained from their design that other designs do not offer. Whether we understand it or appreciate it is another matter.
 
If we are willing to assume that Chord DACs are technically superior, we can ask ourselves why don't everyone prefer them?
 
I think a major challenge is that we are used to hearing DAC chip sound, and for the lucky few, we get to hear and accustom ourselves to DSD/R2R sound. So unless we go to a lot of live acoustic concerts, we bias ourselves to our DAC's sound. Moreover, most of our electronics are then tailored to optimize the sound we get from the DACs we own. If I own a DAC with a DAC chip design that has too much noise floor modulation, making it too bright sounding, I might get a tube preamp/amp to soften the sound or I might get speakers/headphones that roll off in the high frequencies. So a Chord DAC is going to sound too "soft" in my system. If I have a DSD DAC that's slightly soft, I might get an older class D amplifier or my USB/coaxial source might have more RF that gets injected into the DSD DAC to add more noise floor modulation to brighten things up. Of course, the more we deviate from neutrality, the harder it is to recover as much as we possibly can from the original music.
 
I think this is why whenever I read any reviews, I always try to find out what equipments the reviewers are using with the DAC they are testing.
 
And truth be told, nobody likes to be told the gear they own are not neutral or inferior or problematic. But I think a few people have said in this forum that everything we own have some sort of distortion. I certainly think that's true for my own gear and they are a set of compromises in terms of price, space, etc. At the same time, I don't think it's fair to say that since all DACs have distortions, therefore if I like my DAC more than a Chord DAC, Chord DACs are just one of many to choose from. I think the opinion that one prefers another DAC to Chord DAC is valid. But I appreciate people like Romaz who defends Chord DAVE on a sonic and technical merit and correlate the two to explain to people what more they can get out of their music due to the technical advantages of Chord DAVE.
 
With all that said, people buy what they like. So if they still like another DAC over Chord DAVE, good for them.
 
Oct 16, 2016 at 7:52 AM Post #5,151 of 27,054
Relic
'Imaging' is an interesting subject in itself. As you say, speaker placement and decent quality timing of the type produced by Dave Dac helps too. However, I have always been intrigued as to why Conga's, classical guitar, wood blocks for instance image particularly well on almost any old system. At the same time I am sure we have all noticed that anything in the chain which accentuates greater weight to instruments seems to improve the perceived image of the instrument. Also I compared a flat response Kef speaker 205/2 reference and the Sonus Faber Olympica III and noticed that the Olympica III was able to to throw a clear image even when I got close to one of the speakers. It did not give away the real source of the sound until very close. The flat response kef gave away its true source earlier as I moved toward the speaker. The Olympica has a 3-4db frequency dip between 1-3khz and I wondered if this dip helped my brain clarify its reading of positional information in the lower frequencies. This may be a reason for SF's longstanding taylored frequency band dip in this area. I believe the design trick works for them because it effectively narrows the predominant frequency band of 'all' instruments and voices, making it easier for our brain to pinpoint the image without the need for an area of the spectrum which it can do without in relation to calculating position. That 1-3khz frequency band serves to cloud the frequencies which the brain really wants to focus on in order to place the image. Of course there is more to the pleasure of hifi than pure imaging and this drove me to try the Piega Coax 90.2 which is as flat as a millpond in regard to their frequency signature. So I ran them in and noticed that the image greatly improved when the woofers were run in. It was quite noticable. Being an Analyst by nature and profession I couldn't help but spend a little time researching this. I started with what I consider the best imaging instrument, Congas. They tend to fit into a 400hz, 700hz, 900hz peaks sound signature. They have the percussive weight around 400hz therefore. (Depending on the size of instrument this can dip as low as 200hz.) This is interesting because the Piega Coax tweeter/mid crosses over to the woofers at 400hz. I also found a NASA research paper on phase and our ability to place images and coincidentally their tests focussed on the frequency band 200-400hz.

In my view it is the percussive weight in that narrow frequency band which provides easiest (and perhaps most desirable) clue for our brain to do its work. The less prominent that area is, the harder it is to place exactly. All the instruments which image best also seem to exhibit a fairly narrow 'concentrated' bandwidth or have a bump in that frequency range.

This makes me wonder too if Robs finding that it took -340db design to place a church organ 200 ft away. What he had perhaps unwittingly designed was one of the hardest tests in hifi. A massively wide bandwidth instrument in the most notorious of reberating large rooms. Under those circumstances I would expect the 200-400hz frequency band to be all but buried in sound reverberation across a wide frequency range. It's a great test!

Anyway, if I am correct in my assumptions the moral would seem to be, if you are a lover of imaging then probably don't buy a speaker with a dip in the 200-400hz frequency band. :-)


Edit: when I talk about the ease by which the brain is able to calculate placement I am assuming there is no inherent phase distortion when making comparisons.
 
Oct 16, 2016 at 8:00 AM Post #5,152 of 27,054
Sound quality does not matter.

By this I mean it is not the most important thing - its the ability to get emotional satisfaction from music (musicality) that is primary.

And I get to "enjoy" lots of über DAC's at shows - and for me - and of course this is highly personal - they fail big time as regards musicality. I get more musical satisfaction from little old Mojo than any of these über DAC's do.

Moreover, I am absolutely convinced that Mojo (ignore Dave for the time being) will beat any other DAC at any price point when it comes down to enjoying music. Why am I so convinced? Because there are sound technical and objective reasons why this would be the case. Let us consider the three most important things that is important for a DAC/amplifier:


1. Timing - if you spend any time researching psycho-acoustics you will appreciate that timing is the most important parameter that the brain uses to process the data from the ears. It is used for location, timbre, and being able to perceive the starting and stopping of notes. So when I talk about timing problems, what do I mean? Now I am NOT talking about phase errors, or timing differences with frequency response - these are linear errors, and are inconsequential - the brain is used to dealing with these kinds of problems. What I am talking about is when the timing of transients varies either with amplitude (a transient will have a different time depending on whether it is small amplitude or a large amplitude signal) or when the timing of a transient depends upon when it is sampled by the ADC. So if a transient crosses through zero half way between samples, differing interpolation filters will not recreate the timing of the transient accurately, and this sampling dependent uncertainty is highly audible. So how big is the problem of timing? Looking at all the many listening tests I have done, my contention is that a DAC/amp must be accurate down to tens of nS in that surprisingly small timing errors are very audible. 
So what DAC's have minimal timing problems? Let's look at the interpolation filter first, something that is in all DAC's (yes even NOS DAC's have an interpolation filter). Now the job of the interpolation filter is to reconstruct the timing of transients and to remove the HF images of the signal that extends to infinite frequencies that is due to the sampling process. In short, it is part of the process that takes the discontinuous digital data and converts it back into a continuous analogue signal. The better the interpolation filter, the more accurate this is done (closer to the original analogue signal in the ADC before it was sampled). Now the accuracy of the filter in terms of how well transient timing is reconstructed depends upon the filter tap length, and the type of filter. NOS filters are by far the worst, with timing errors of up to a hundred uS. A NOS filter (this is a bad term, all DAC's oversample) is actually a zero order hold interpolation filter, where the oversample rate is determined by the DAC that is used. Because the timing errors are so huge, they sound very soft (when you get timing errors, the brain can't deal with the data, so it can't actually perceive the starting and stopping of notes - and if you can't hear the starting of a note, it sounds soft or out of focus). Now some people like this; but it is clearly artificial and is categorically not transparent. The usual FIR filter is a half band interpolation filter, with a tens of taps. This filter is better than zero order hold (NOS) for timing, but still has massive errors, as there is considerable measurable sampling image errors. These filters return the original sample unchanged, and they are cheap to implement which is why they are used for 99% of the time. Apodizing filters can offer better performance, but they still have substantial timing errors.
So how can you reconstruct the timing of transients perfectly? If you look at sampling theory, it has been proven that a Whittaker-Shannon interpolation filter will return a bandwidth limited sampled signal absolutely perfectly; there will be no timing or amplitude errors at all. But to have an ideal interpolation filter you need a sinc impulse response and many many taps; to absolutely guarantee just 16 bit performance under all circumstances the coefficients need to be smaller than 16 bits - and this happens at about 1 million taps for a 16 FS filter. And every listening test I have ever done gives the same conclusion - more taps, better and more transparent sound quality. Even with Dave at 164,000 taps, we have not reached the subjective limit yet.
So onto the next timing problem - amplitude errors. Now R2R DAC's have slow FET's to switch the resistors in and out, and these themselves cause their own glitch issues. Moreover, they are very slow - you can only go to 16FS max - so that means the timing is limited to only 1.3 uS. DSD dac's have very large amplitude timing errors too - a small signal has a much bigger delay than a large signal. Currently, the only technique that has the smallest amplitude related timing error is pulse array, as it runs 5 bits at 104 MHz - many times faster than any other DAC topology.
So to solve the timing issues you absolutely need extremely fast DAC's, and very large tap lengths. Even Mojo easily beats all other non Chord DAC's in this regard.   

2. Small signal resolution. Clearly small signals are vital - if you can't hear the tiny details, then it no longer becomes a believable performance. Small signals also are used for depth perception, something that I am personally very interested in. Go to a cathedral and listen to an organ at 100 m away; it sounds 100 m away - but reproduced audio is severely depth truncated. Now small signal linearity is measured using fundamental linearity tests - you measure the amplitude of a -60db, set that value as a reference, then measure at -120 dB say. It should be exactly -120.000 dB, but a real DAC won't be. Delta sigma or DSD will actually attenuate the level, R2R will have random errors due to resistor tolerance problems. When it comes to depth perception, there is something extremely strange - any small signal amplitude error (no matter how small) affects depth perception. That's why Dave has 350 dB performance noise shaping, as increasing the performance of the noise shaper gave much better depth performance. Now R2R DAC's have easily measurable small signal errors; DSD is only -120dB accurate noise shaping; conventional DAC noise shapers are about -140dB; Mojo is at -200 dB, and Dave takes the record at -350 dB. I have published FFT's showing Dave's noise shaper performance.
Now this issue is more complex than this; you also need simple analogue stages to further improve performance, you additionally you need a distortion performance that gets better as the signal get smaller.

3. Noise floor modulation. This is another can of worms, as DAC's have a multitude of problems from different areas for this. Noise floor modulation is subjectively very important; it is when the noise floor changes with signal level. All conventional DAC's have large amounts of noise floor modulation, and it is my contention that it is very audible - it adds a hardness or grain to the sound. Even the tiniest amount affects smoothness and refinement. With Mojo, we have no measurable noise floor modulation - and similarly Dave but with an even lower noise floor. I have not seen any other DAC come close to this performance.

This gives you a brief flavour of the issues involved in producing a truly transparent DAC; and for me it's only by having a truly transparent DAC that one can get musicality. But of course some people have very different tastes, and respond differently to music. Some people like a particular sound; I have done some listening sessions when I drew the complete opposite opinion to somebody else. Some people like distortion; others like an overly soft warm sound. Whatever floats your boat I guess.

Rob

Fully agreed on the importance of musicality and the emotions we get from hearing music.
Let's also agree that having a truly transparent DAC is the best way to get musicality.
But than my next point is the following: if transparency guarantees the exact reproduction of whatever the sound engineer wanted to render during the recording session, the journey to musicality nirvana is still long. We should than start questioning if the recording itself is even capable of delivering such emotions.

It's a difficult point I'm trying to make here. What I mean is that I am not convinced that it's even realistic to strive for a full chain of perfection, because even a recording room which absorbs all sound reflection can actually sound very weird and unnatural.
Of course the capability of reproducing little details and other points you mentioned are essential to make music sound "realistic".
However I am more keen to believe that it's a combination of little errors and deformations that can generate that magic harmony that eventually makes you feel the emotions.

Sorry, I don't know if this makes any sense, I am not an expert at all but I can definitely tell what gives me that "click" when I hear a DAC.
 
Oct 16, 2016 at 8:31 AM Post #5,153 of 27,054
Can you guys help and
comment your impression on how your system compare with the sound in the video. Would love to hear opinions from TOTL headphone users. ​
 
Go to the video in this link(https://www.youtube.com/watch?v=u2AJeC7R_no) and go to 8:44. Listen the test song that the couple use to test the speaker. Afterwards, listen to the same song in your system. The name of the song is 
Zigeunerweisen, Op. 20, No. 1 by "Pablo de Sarasate, Lara St. John, Ilan Rechtman". The same cut start from 6:22 into the song. You can search using Tidal/other streaming service. ​
 
Oct 16, 2016 at 8:41 AM Post #5,154 of 27,054
Jawed, have you compared the Smyth Realizer A16 to Darin Fong's Out of Your Head software?

I can't compare them since I've only heard A16 at CanJam.

When I got home after that day at Canjam I tried OOYH. I wasn't impressed.

From the demos on their website, the Individual Virtual Speaker Demo - 8 speakers fails to provide a convincing image of the left, front and right speakers. It could be failing entirely because the "average" HRTF is a terrible match for me. A16 was utterly convincing doing the same thing. A16 does scale and space, but OOYH with the default configuration, for me, presents a fake, vaguely spacious sound.

I think there is nothing for it but to have your own HRTF measured with microphones or computed. There is a lot of research on computing HRTFs and it appears likely that computed HRTFs will be good enough.

OOYH can render a pleasingly spacious sound from stereo or multi-channel sources. I don't like the sound of speakers in a room as boom and wall reflection effects are unpleasant. So hearing those faults spoils the experience for me. Both OOYH and A16 are based on speaker sound in a room, so both suffer from this fault in my opinion. The virtual speaker demo I referenced earlier appears to be trying to model a bad sounding room.

My preference is for an idealised reproduction model where the boom and wall reflection effects in playback solely derive from the recording. The space you perceive should be solely the space in the recording, devoid of any artefacts from the listening environment.

I have no idea if OOYH with a personalised measurement can be as good as A16. For many people wanting to use A16 they're going to be stuck using an approximation: combining a measured room/speaker system uploaded by someone else with their personalised measurement. That wouldn't be as good as taking your personalised measurement while in that room with that speaker system. I don't know how much of a compromise that would be.
 
Oct 16, 2016 at 9:27 AM Post #5,155 of 27,054
I would like to seek your advice/opinions on streaming red book as opposed to top quality CD disc players. I currently listen exclusively to CD played through a Red Reference III digitally linked to Dave. My question is, have any of you guys compared top quality CD replay against a streaming setup playing red book from a solid hard disk into Dave via USB?

Tia
 
Oct 16, 2016 at 10:51 AM Post #5,156 of 27,054
I have no idea if OOYH with a personalised measurement can be as good as A16. For many people wanting to use A16 they're going to be stuck using an approximation: combining a measured room/speaker system uploaded by someone else with their personalised measurement. That wouldn't be as good as taking your personalised measurement while in that room with that speaker system. I don't know how much of a compromise that would be.


It's been said by Smyth Research that non-personalised HRTFs Will give approximately 90%+ of the experience, but not 100%.

I was hit and heavy for an A16 until I realised that I wouldn't be able to get my hands on a new one until late 2017, at best. Chances are 2018.
 
Oct 16, 2016 at 10:54 AM Post #5,157 of 27,054
I would like to seek your advice/opinions on streaming red book as opposed to top quality CD disc players. I currently listen exclusively to CD played through a Red Reference III digitally linked to Dave. My question is, have any of you guys compared top quality CD replay against a streaming setup playing red book from a solid hard disk into Dave via USB?

Tia


My take is that the sonic difference to me playing off a cheap Oppo BDP-103 via Toslink into the DAVE vs my USB setup (which is not as good as microRendu I presume) is present but not very significant. So for someone who is comfortable with playing CDs exclusively and has never done computer audio, I'd probably would not recommend anyone to switch from a CD transport to computer audio. The other way to think about it is if you're going to go into computer audio for DAVE, you're probably going to go all in and get the microRendu so the entire setup process is a bit intense if you're computer audio naive.
 
Besides, I thought Chord was planning to upgrade the Red III so that you can still use the Red III to upsample to 176kHz 24-bit but now with even more taps than the DAVE can provide. That could definitely be a worthwhile upgrade.
 
Anyway, my take is if you want to go into computer audio, you should just take the dive and have fun with it. But I wouldn't get into computer audio just to squeeze out that extra bit of performance out of DAVE. It'll be interesting to see what others think.
 
Oct 16, 2016 at 11:18 AM Post #5,158 of 27,054
Thanks for your honest reply. Obviously you are tempting me with stories of an RRIII upgrade. Oddly though I currently find 44.1 on the RR3 more natural going into Dave than 88.2 there is an ease to it that the tighter 88.2 rendition does not seem to posess.
 
Oct 16, 2016 at 12:15 PM Post #5,159 of 27,054
Thanks for your honest reply. Obviously you are tempting me with stories of an RRIII upgrade. Oddly though I currently find 44.1 on the RR3 more natural going into Dave than 88.2 there is an ease to it that the tighter 88.2 rendition does not seem to posess.


I think Red III currently has 18000 taps to do the 4fs upsampling. If you send 44.1 to DAVE, with 164000 taps, I'm guessing 41000 taps are devoted to the first 4fs upsampling. That's why sending 44.1 to DAVE sounds better. Besides DAVE has a more up-to-date WTA algorithm. I'm simplifying and misrepresenting a few details and guessing a bit but you get the gist. I'm sure Rob Watts and John Franks can fill you in on more details that's actually correct instead of my guesses. Not sure how many taps go into the Red III upgrade...
 
Oct 16, 2016 at 5:20 PM Post #5,160 of 27,054
Thanks for the reply Jawed

:beerchug:
 

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