Intriguing list....my experience with the Vivaldi DAC is that using HQ Player is a wash....
Do better DACs benefit from upsampling the source (HQPlayer)? Does HQPlayer just change the sound?
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Intriguing list....my experience with the Vivaldi DAC is that using HQ Player is a wash....
Gentlemen, I appreciate the discussion.
Here's a quote I found from a guy named Watts, who designed the Chord DAVE dac, which competes with any DAC in the market today, regardless of price tag.
"Oh dear. Do NOT use your computer to up-sample or change the data when you use one of my DAC's.
All competent DAC's up-sample and filter internally; the issue is how well that filtering is done, in terms of how well the timing of transients is reconstructed from the original analogue. Computers are poor devices to use for manipulating data in real time as they are concurrent serial devices - everything has to go through one to 8 processors in sequence. With hardware and FPGA's you do not need to do that, you can do thousands of operations in parallel. Dave has 166 DSP cores with each core being able to do one FIR tap in one clock cycle. That is incredibly powerful processing power way more powerful than a PC.
But its not just about raw processing power but the algorithm for the filter. The WTA filter is the only algorithm that has been designed to reduce timing of transients errors, and the only one that has been optimised by thousands of listening tests.
So the long and the short is don't let the source mess with the signal (except perhaps with a good EQ program) and let Mojo (or DAVE) deal with the original data, as Mojo (or DAVE) is way more capable"
Does any one understand the technical rationale behind what he is saying?
Here's a quote I found from a guy named Watts, who designed the Chord DAVE dac, which competes with any DAC in the market today, regardless of price tag.
"Oh dear. Do NOT use your computer to up-sample or change the data when you use one of my DAC's.
All competent DAC's up-sample and filter internally; the issue is how well that filtering is done, in terms of how well the timing of transients is reconstructed from the original analogue. Computers are poor devices to use for manipulating data in real time as they are concurrent serial devices - everything has to go through one to 8 processors in sequence. With hardware and FPGA's you do not need to do that, you can do thousands of operations in parallel. Dave has 166 DSP cores with each core being able to do one FIR tap in one clock cycle. That is incredibly powerful processing power way more powerful than a PC.
But its not just about raw processing power but the algorithm for the filter. The WTA filter is the only algorithm that has been designed to reduce timing of transients errors, and the only one that has been optimised by thousands of listening tests.
So the long and the short is don't let the source mess with the signal (except perhaps with a good EQ program) and let Mojo (or DAVE) deal with the original data, as Mojo (or DAVE) is way more capable"
Does any one understand the technical rationale behind what he is saying?
Furthermore, why hasn't a digital company come out with a "total solution" that involves upsampling via a computer before sending the Information to the dac. After all we see companies come out with their own servers, clocks, etc., and designers find al kinds of ways to improve the sound/ upsell their stuff.
A lot of money is involved in a sale to even one customer. Guys ain't dumb. So why haven't we seen more of this approach in the free market?
A lot of money is involved in a sale to even one customer. Guys ain't dumb. So why haven't we seen more of this approach in the free market?
I think there is a marketing bull around up sampling TBH, it is selling DACs and has done that for 20 years. Lately there is an influx of R-2R DACs that deal with the data as bit perfect. This avoids the whole Delta Sigma 'issue' and subsequent guess work and in some cases the filtering required. I have tried up sampling with R-2R DACs and all it does is smear the sound, flatten the stage and depth and basically bleach out the music.
I am talking only about PCM, not considering DSD conversion.
Rob is talking about his FPGA DACs which, like dCS no doubt prefer the data as supplied. I can see that would work, though I haven't heard the DAVE yet. Oddly though, many 20K DACs are still doing major up sampling still (CH Precision for example and all MSB models except the Select DAC I have been told) even though they are R-2R DACs. I am not sure why or what it achieves. One of the advantages may be it 'looks better' on the scope and lab results, but does it sound better?
Correction: MSB doesn't upsample, unless you tell it to. Analog DAC and SELECT DAC don't upsample at all.
WRT to multi - core processors as "concurrent serial devices" that was much more of an issue or said more explicitly a bottleneck years back. Today, multi - core scheduling has improved greatly. This is a primary reason why computers have been getting more responsive over the last 10 years while frequencies haven't budged. Leverage more cores, optimized scheduling along with increased peripheral device and memory speed. Net is - I don't agree with Watts.
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Sorry for confusion. What I mean is that we don't see an "end to end" solution that includes a separate computer on the front end. Some of the best digital Companies like naim, totaldac, msb, esoteric, etc., all seem to be happy to upsell a clock or a server. But not a separate up sampling computer on the front end...
Phasure produces a NOS DAC that requires software upsampling through XXHighEnd, also written by them. Some users employ HQPlayer instead.
I read through their technical paper and it is quite an impressive design. In a nutshell, he implements all upsampling and noise shaping using very long filters (i.e. closer to ideal response) using parallel arrays of DSP inside the FPGA. While HQPlayer has a chinese menu of such algorithms, they say that they have picked one and optimized/implemented that.
The measurements in their technical documentation are exemplary. Depending on what you did in HQPlayer, you could indeed add distortion products that would not be there (to the extent the filter response is not as good as what they have in hardware).
The above implicitly answers the "why HQPlayer." Because it is implemented in software, you can play with different algorithms whereas with hardware you are "stuck" with the one that is inside the DAC. Personally I see no reason for the complexity that HQPlayer brings, nor do I believe in subjective improvements people here. So that advantage is neither here, nor there in my book.
Give it time, the realization that external control of filters and upsampling and reported benefits seem to be fairly new.
I think that they do not see the market for it in their segment and I agree. All too many "audiophiles" express disdain for having a computer in their systems. They seem to prefer be fooled by having them camouflaged into what passes for audio componentry.
"the only algorithm that has been designed to reduce timing of transient errors" A linear phase filter? I think I've seen them provided by others.
Here's the impulse response of the Chord Dave courtesy of JA @ Stereophile.
If someone wants to hear similar from HQPlayer the filter is called poly-sinc-xtr. I suppose it would be debatable to which sounds better through a Dave but it gives many others the opportunity to get Dave like sound from their own DAC if they have the processing power.
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my understanding of 'Ravenna' is that it's a proprietary version of Ethernet used by Merging Technology. likely I don't have that quite exactly right but that is what I recall from when I was investigating the NADAC.
Ethernet as I understand it is where the dac has it's own processor for directly accessing files over a network, instead of accessing a server for the data.
and some higher level dacs allow you to add an Ethernet hub to the dac, as opposed to using USB. one typical negative of Ethernet is the noise from the processor inside the dac. and OTOH it's said that a USB-server approach adds an unneeded step to the process.
be gentle if I've got the techie stuff a bit skewed......I'm a user/listener and not an authority on the nuts and bolts, and typically leave the heavy lifting to others.
I emailed MSB last year and was told the Analogue and Diamond upsample on the fly at 80bit. Only the Select does not.
Not all
But I am seeing the same thing. Even when I first ventured into streaming I hated everything that wasn't Linn due to the requirement to use USB with terrible apps for control forcing you to have a monitor and mouse in your audio room. That time has passed and we have Roon now.
Yes, the Phasure NOS1 DAC employs four PCM1704U-K chips per channel and is a non-oversampling and filterless design. It is capable of accepting a 705.6/768 rate signal via USB. It is assumed that the redbook/hi-rez signal will upsampled to these rates in the software player. What's the advantage of this? Well, being able to utilize the massive resources at hand in an audio PC to do the upsampling and noise-shaping. Rob Watts states that his latest WTA filter in the DAVE uses 164,000 taps. I can use the 'closed-form' filter in HQPlayer with massively more taps. (Although I have to say that if I were using a DAVE, I probably wouldn't bother upsampling in the software player.)
I have two NOS1 DACs set up - one in my main system using XXHighEnd, and the other in my office system using HQPlayer (integrated with Roon). Both work superbly well. Although I haven't had a chance to compare against many other high-end DACs, the Phasure NOS1 DAC was 'responsible' for my selling my Pacific Mircosonics Model Two.
Mani.
Looks to me he's spouting a fair amount of nonsense, or at least muddying the waters with irrelevancies.
Computers do not need to handle audio 'in real time' - they operate so much faster than the sample rate they can do batches of processing and buffer it up for later delivery to the DAC. So 'in real-time' here is a distraction. True enough that FPGAs do DSP in parallel but there's no inherent advantage to that other than the ability to run much slower clock rates.
He could just say something like 'Our filter is optimized to work best with our particular hardware, if you use an external upsampler you're taking a risk that its not going to sound as good as ours'. No need for all the fluff about parallel vs serial.
ROON is good for many people but the issue is that proprietary hardware, such as audiophile servers/players, are constrained in two important ways. First, they usually employ the least capable internals necessary to perform the job as seen by the manufacturer at the time of design. There's usually no future upgrade option when/if there are increases in demand as new features and formats roll out. Second, they are constrained by limited ability to be reprogrammed and/or load entirely new software in response to demand.Not all
But I am seeing the same thing. Even when I first ventured into streaming I hated everything that wasn't Linn due to the requirement to use USB with terrible apps for control forcing you to have a monitor and mouse in your audio room. That time has passed and we have Roon now.
And this is just speaking about those that do not include DACs. For those that do, they are virtually defined by their hardware.
This is why I am a big fan of server/players based on general purpose computers (Win, Mac or Linux), even if they must be packaged to look like an audio component and have a limited GUI in order not to create anxiety in computer-phobic audiophiles.
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