Audirvana Plus 3 (official thread)

[Speed Racer]

2 minutes ago, jhwalker said:

 

Exactly - there is no such thing as "real time" processing (maybe when we get to quantum computing). 

 

Processing takes time, and it's code that does it - whether the code is in a chip (hardware) somewhere, or loaded in software, or . . . as long as the processing takes place ahead of playback needs, the difference between "real time" (misnomer anyway) and "a very short time" is inconsequential.  And computers indisputably have more flexibility for updated filters, etc., than can be encoded in a chip.

 

Wow. You are wrong. Of course there is a such a thing a "real-time" processing. You guys need to step back and learn about real-time processing before you say that it does not exist.

[Innocent Bystander]

2 hours ago, astrostar59 said:

I disagree. A Mac Mini is £500 approx. And add the Uptone Audio MMK at £130 and a Linear PS at around £150 and you then have a very user friendly, updatable and compact solution. I get the music server pre-built ideas, but I much prefer to be able to tweak and update a computer.

Yes with secondhand late 2012 MM, the price can be much lower as I hinted (my own is 2012, bought secondhand, which I upgraded to 16GB plus 2x 1TB SSDs, and could easily upgrade the SSDs further if I wished). But if you're talking new MM, it is not readily tweakable or updateable on the hardware side, and costs £180 more than your figure for the 2.5GHz version, with ITB HDD and no option for larger HDD, plus of course the cost of A+, so a bit over £1K.  If you want more storage, an additional 3GB external thunderbolt drive to match the gross capacity of the Melco N1A (4TB) would take that closer to £1300, and quite a bit more if you wanted equivalence to the Melco with 2x2GB raid storage on the Mac bus. That is where I was coming from in my comment in terms of Melco not a lot more, and sufficiently close for a serious consideration of pros and cons of each.

 

Personally I don't want to keep tweaking, but want to find the best sound and don't mind tweaking a bit to get there. I am very happy with sound quality from Mac Mini and A+, but it is a real hassle to use because of the library limitations, and impossible for my non-techie better half to cope with, so she can't play any music when I'm not there, which is rather sad, so if my MM died I would seriously consider alternatives. But hopefully the library will be fixed and all will be wonderful - apart from Apple's annoying lockdowm of its hardware.

[mansr]

18 minutes ago, Speed Racer said:

 

I can't help that you don't understand the difference between real-time and high-performance systems.

 

You seem to believe that a system can't be both. Granted, providing real-time guarantees often comes with a cost in (peak) throughput, but for audio playback on a modern PC that's not an issue. In fact, the demands of audio are low enough that you don't even need an RTOS for reliable operation. Just how do you think any music player works if software can't do things in real time?

[mansr]

27 minutes ago, jhwalker said:

 

Exactly - there is no such thing as "real time" processing (maybe when we get to quantum computing). 

 

Processing takes time, and it's code that does it - whether the code is in a chip (hardware) somewhere, or loaded in software, or . . . as long as the processing takes place ahead of playback needs, the difference between "real time" (misnomer anyway) and "a very short time" is inconsequential.  And computers indisputably have more flexibility for updated filters, etc., than can be encoded in a chip.

 

Real-time in computing doesn't mean zero latency. It means bounded latency. In other words, the processing can produce output with a fixed delay for indefinite periods of time.

[Speed Racer]

22 minutes ago, mansr said:

 

You seem to believe that a system can't be both. Granted, providing real-time guarantees often comes with a cost in (peak) throughput, but for audio playback on a modern PC that's not an issue. In fact, the demands of audio are low enough that you don't even need an RTOS for reliable operation. Just how do you think any music player works if software can't do things in real time?

 

If what you say were true, then Audirvana would have no need for its "SysOptimizer". So please stop with the misinformation. By definition, "real-time" means in cannot be interrupted by anything other system requirement. PC's do not run user programs in real-time.

 

My point is that DACs that upsample have all the processing power they need to upsample in real-time. They don't need any more processing power than they have to process the data in real-time and they are not somehow inferior to computer systems that have more processing power.

[mansr]

16 minutes ago, Speed Racer said:

If what you say were true, then Audirvana would have no need for its "SysOptimizer". So please stop with the misinformation. By definition, "real-time" means in cannot be interrupted by anything other system requirement. PC's do not run user programs in real-time.

 

And yet our music players work just fine.

[Speed Racer]

12 minutes ago, mansr said:

 

And yet our music players work just fine.

 

Yes, the vast majority of the time. Yet an upsampling DAC works all the time every time. That the difference between "real-time" and "high performance".....

[cdx777]

On 3/14/2017 at 11:51 PM, RunHomeSlow said:

First time tonight i turn on 3.02 and i got a error message for the first time ever with System Optimizer...

something like: not all part might be working... ''broken pipe'' something. Close A+ and reopen... no message. Green light is lighten.

 

While continuing writing this message, my song seems a bit strange... thought it was skipping a note.... then 10 sec later maybe was not moving in right direction if i can say that... then A+ Crash.

 

Damien, i did a restart of MAC and this time A+ start first shot without message for SysOpt... maybe something with some old cache like you said in another thread, that the restart cured, i don't know. All seems working now. First bug in a long Time with A+. That song playing was Jimmy Smith SACD DSF file, FYI.

 

 

 

I have the same problem but with 3.0.3 since I updated. it displayed the error message for the first but now it just crashes and quits whenever I play a song... unless my library is syncing, it asks me if I want to terminate if i say cancel it keeps playing music and doesn't crash. it's weird the only way to keep it running is have a ongoing playlist that doesn't stop or when it's syncing my files... can somebody please help thank you?

obviously I tried uninstall/reinstall, delete the preferences and other files... nothing works...

 

[cdx777]

On 3/16/2017 at 4:58 AM, RunHomeSlow said:

...

 

running into the same thing... did this get a fix?

[RunHomeSlow]

That was introduce for me with 3.0.2 (Damien did something with the sysOpt...arf!) all was fine with 3 and 3.0.1.

3.0.3 randomly displayed that message, but don't crash since i reinsalled 3.0.1, tested it, then reinstalled 3.0.3 directly no 3.0.2... When it occured, i don't play that song even if it will. I just quit and reload A+ and usually the message is not reappearing...

[Jud]

2 hours ago, Speed Racer said:

 

Yes, the vast majority of the time. Yet an upsampling DAC works all the time every time. That the difference between "real-time" and "high performance".....

 

Yep, DACs are real-time, that's why they don't use buffers.

[Jud]

3 hours ago, Speed Racer said:

 

If what you say were true, then Audirvana would have no need for its "SysOptimizer". So please stop with the misinformation. By definition, "real-time" means in cannot be interrupted by anything other system requirement. PC's do not run user programs in real-time.

 

My point is that DACs that upsample have all the processing power they need to upsample in real-time. They don't need any more processing power than they have to process the data in real-time and they are not somehow inferior to computer systems that have more processing power.

 

SysOptimizer isn't about real-time, it's about priorities in a system that would otherwise be devoting resources to doing several more things "at once" from the user's perspective.  It doesn't shut down everything else.

 

Nor is filtering in a computer vs. a DAC about real-time resources.  DACs built with chips boast of 13,000-tap filters; 10,000 taps is the *lowest* setting possible with iZotope SRC on A+, and a couple million are possible.  Not that number of taps is the be-all and end-all, just that it gives some indication of the extra freedom provided filter designers working on a computer vs. a chip.  I can also adjust a half-dozen or so filter parameters over a wide range to my personal taste with A+; how many filter settings are possible with your DAC?  How often is your DAC chip's firmware updated?  Damien has updated A+ several times in the past few weeks, and I can change filtering or sigma delta modulators in the next 5 minutes if I'd like.

 

So this has much more to do with more opportunities to work with greater resources for filter designers and more choices for consumers than whether filtering will be applied in "real time."

[audiocanyon]

8 hours ago, Jud said:

 

It will be interesting to see whether you even think you hear a difference.

 

Edit: You may want to try upsampling to DSD128 and see what that sounds like to you, if the Benchmark will accept that; if not, you can try DSD64.

 

Further edit: DSD will initially be lower volume, I believe.  So for any comparison you would want volumes equalized.

 

I initially was upsampling to DSD64, the highest the Benchmark DAC2 will accept.  Recently I tried maximum sample rate upsampling and feel it adds a bit more space and clarity in the sound stage. My DAC reports 192, higher than with DSD64. 

 

It seemed that DSD64 was richer than my DAC alone with A+ 2.6.  I haven't tried my DAC without any upsampling yet with A+3. That's my next experiment.

[Speed Racer]

40 minutes ago, Jud said:

 

Yep, DACs are real-time, that's why they don't use buffers.

 

I can't argue with the unarmed.....

[Speed Racer]

25 minutes ago, Jud said:

 

SysOptimizer isn't about real-time, it's about priorities in a system that would otherwise be devoting resources to doing several more things "at once" from the user's perspective.  It doesn't shut down everything else.

 

Nor is filtering in a computer vs. a DAC about real-time resources.  DACs built with chips boast of 13,000-tap filters; 10,000 taps is the *lowest* setting possible with iZotope SRC on A+, and a couple million are possible.  Not that number of taps is the be-all and end-all, just that it gives some indication of the extra freedom provided filter designers working on a computer vs. a chip.  I can also adjust a half-dozen or so filter parameters over a wide range to my personal taste with A+; how many filter settings are possible with your DAC?  How often is your DAC chip's firmware updated?  Damien has updated A+ several times in the past few weeks, and I can change filtering or sigma delta modulators in the next 5 minutes if I'd like.

 

So this has much more to do with more opportunities to work with greater resources for filter designers and more choices for consumers than whether filtering will be applied in "real time."

 

Since Audirvana does not operate in real-time, it needs to modify the system so that it is less likely to be interrupted.

 

The filtering in a DAC has to operate in real-time so it always has enough power to do what i needs to do in real-time.

 

I never argued that you couldn't do more filtering in the computer. I just argued that you saying a computer has more computing power does not necessarily mean anything.

 

If a DAC manufacturer wanted to implement more complex filtering, all they have to do is add the computing power to do so. Again, in real-time, which is much different than "high performance" like you get a with a conventional computer. When something goes in the DAC, it has to come out on time every time.

[RunHomeSlow]

40 minutes ago, Speed Racer said:

If a DAC manufacturer wanted to implement more complex filtering, all they have to do is add the computing power to do so. Again, in real-time, which is much different than "high performance" like you get a with a conventional computer. When something goes in the DAC, it has to come out on time every time.

 

Don't want to enter that discussion, cause i dont have that knowledge, but curious about all the troubles i had before.

 

What about like with A+ at the time i tried without sucess to upsample pcm to the max my dac can do (352/384) and it was sometimes just screeeech sound or other times the dac wouldn't stick on the sample rate or skip notes... was that the software fault or the dac that couldn't come out something every time ?

 

since A+ is able to upsample to dsd now at my dac dsd128 max all my problems are gone with that. Still using the same dac !?!

[RunHomeSlow]

I can say also that at the time i changed my mac mini to this late 2012 with 16gb and still had trouble upsampling like said up there pcm 44 to 352..,

[Jud]

1 hour ago, Speed Racer said:

 

Since Audirvana does not operate in real-time, it needs to modify the system so that it is less likely to be interrupted.

 

Audirvana worked fine for years before Damien made SysOptimizer available.

 

1 hour ago, Speed Racer said:

If a DAC manufacturer wanted to implement more complex filtering, all they have to do is add the computing power to do so. Again, in real-time, which is much different than "high performance" like you get a with a conventional computer. When something goes in the DAC, it has to come out on time every time.

 

"All they have to do is add the computing power...."  With which specific DAC chip made today that will equal the resources available from a modern CPU?

 

I have no idea where this obsession with "real-time" comes from.  If we are talking about a USB DAC and A+, the software-filtered bitstream is stored in RAM, then sent when the user wants from the computer.  It is the subject of communication between the DAC's USB receiver and the computer, which sends packets in order to keep the DAC's buffer filled.  These stages are not real time, regardless of whether you upsample in A+ or leave it to your DAC.  If the bitstream is upsampled it skips some or all of the DAC chip filtering before the final analog filter (just as it would if the file were originally hi res).  In whatever part of the DAC's processing it doesn't skip, it behaves exactly the same, for "real time" purposes, as something coming out of the DAC's internal chip.  

 

If you're talking about actual dropouts while the computer is performing software upsampling, that's not a problem for PCM rates.  For DSD rates, it depends on how modern your computer is and what DSD rates you want to get to.  My 8 year old notebook with a Core 2 Duo CPU will do DSD128 without a problem.  For DSD256 I might have to wait for part of a song to load into memory.  But that's if I choose upsampling to DSD256 on an 8 year old machine.

 

So where is it that this "real time" thing produces better sound quality, or are you not talking about quality at all, but just avoiding dropouts?

[Innocent Bystander]

2 hours ago, Jud said:

 

I can also adjust a half-dozen or so filter parameters over a wide range to my personal taste with A+; how many filter settings are possible with your DAC?  How often is your DAC chip's firmware updated?  Damien has updated A+ several times in the past few weeks, and I can change filtering or sigma delta modulators in the next 5 minutes if I'd like.

 

Why would you want to do that? Surely the goal is to find the most accurate conversion of the encoded file of digitally recorded music into an analogue signal, and contantly tweaking suggests it not working. Or do you mean you're effectively using the process as DSP, in effect tone control, to adjust individual recordings to a sound that pleases you? The two are very different, though of course neither is wrong, as it is all about musical enjoyment.

 

in my view the DAC designers do their best, and at least with the upper echelon that means a lot of listening, while renderer designers do likewise, and I was under the impression that A+ is the result of listening not just theoretical, however perhaps I am wrong and it is just coincidence when any given version sounds right with the recommended settings?

[Jud]

1 minute ago, Innocent Bystander said:

Why would you want to do that? Surely the goal is to find the most accurate conversion of the encoded file of digitally recorded music into an analogue signal, and contantly tweaking suggests it not working. Or do you mean you're effectively using the process as DSP, in effect tone control, to adjust individual recordings to a sound that pleases you? The two are very different, though of course neither is wrong, as it is all about musical enjoyment.

 

in my view the DAC designers do their best, and at least with the upper echelon that means a lot of listening, while renderer designers do likewise, and I was under the impression that A+ is the result of listening not just theoretical, however perhaps I am wrong and it is just coincidence when any given version sounds right with the recommended settings?

 

 

I can completely bypass my DAC's filtering with A+.  My speakers have linear phase crossovers.  I have an iFi DAC.  I have an 8 year old computer.  My filter and modulator settings reflect that.  Your system is likely different, and you can have different filtering parameters and modulator choice to reflect that.  What if you change (or want to change) something else in your system and it no longer sounds quite right to you with your current DAC?  I can keep my DAC and change the filtering.  There are discussions about various types of filtering on this forum and elsewhere; I can try them if I like, keep the changes if it's an improvement, go back to what I had if it's not.  I've tried and rejected minimum phase and apodizing filters.  It was a matter of moving sliders.  Other people would have had to buy or borrow entirely new DACs.

 

I could go on, but you get the idea.

 

So what is the result of this experimentation?  For music listening, as opposed to testing, I've used just two different filtering setups in the past three years.

[watercourse]

37 minutes ago, Innocent Bystander said:

Why would you want to do that? Surely the goal is to find the most accurate conversion of the encoded file of digitally recorded music into an analogue signal, and contantly tweaking suggests it not working. Or do you mean you're effectively using the process as DSP, in effect tone control, to adjust individual recordings to a sound that pleases you? The two are very different, though of course neither is wrong, as it is all about musical enjoyment.

 

in my view the DAC designers do their best, and at least with the upper echelon that means a lot of listening, while renderer designers do likewise, and I was under the impression that A+ is the result of listening not just theoretical, however perhaps I am wrong and it is just coincidence when any given version sounds right with the recommended settings?

I don't change settings often, as I don't have that many DSD recordings, but I have enough DSD files to notice that one size does not fit all when it comes to filter settings. Top end energy is not the same IME between DSD and PCM: if you tend to listen to complete albums at a time, you might want to tweak settings based on media characteristics.

Also, you have mentioned "accuracy" a couple of times in posts now, and by that I assume you mean tonal accuracy. I neglected to respond earlier, but I also agree that A+ 2.6.6 is tonally accurate. However, compared to my reference Bel Canto PL1, computer playback still lacked the finesse/realism of more holographic soundstaging, denser images, and relaxed presentation compared to my reference. These factors are difficult to measure or quantify, and sometimes only apparent by A/B comparisons. 

A+ v3 is still a step back from my reference, but a lot closer than v2.6.6. The difference could be down to my computer source, power supply, or other potential factors that I have not fully explored as of now, and not necessarily the software. I may also not yet have hit upon the magic filter combo. But when I don't directly compare to my reference, I don't perceive deficiencies in the same way as I did with 2.6.6.

[Innocent Bystander]

1 hour ago, watercourse said:

I don't change settings often, as I don't have that many DSD recordings, but I have enough DSD files to notice that one size does not fit all when it comes to filter settings. Top end energy is not the same IME between DSD and PCM: if you tend to listen to complete albums at a time, you might want to tweak settings based on media characteristics.

Also, you have mentioned "accuracy" a couple of times in posts now, and by that I assume you mean tonal accuracy. I neglected to respond earlier, but I also agree that A+ 2.6.6 is tonally accurate. However, compared to my reference Bel Canto PL1, computer playback still lacked the finesse/realism of more holographic soundstaging, denser images, and relaxed presentation compared to my reference. These factors are difficult to measure or quantify, and sometimes only apparent by A/B comparisons. 

A+ v3 is still a step back from my reference, but a lot closer than v2.6.6. The difference could be down to my computer source, power supply, or other potential factors that I have not fully explored as of now, and not necessarily the software. I may also not yet have hit upon the magic filter combo. But when I don't directly compare to my reference, I don't perceive deficiencies in the same way as I did with 2.6.6.

Interesting. 

Perhaps my perceptive is a bit different because my DAC does wonderful things, holographic soundstaging and remarkable clarity as fed with untweaked and un-upsampled A2.6.x, with a naturalness that makes me wary of change. I may  try v3 out of curiosity in due course, but I'm in no hurry, and in the meantime I.m hoping Damien will fix the library.

[RunHomeSlow]

People like me (i think) really wants to have the best they could get EASILY from their software or DAC, looking in forums like here on ComputerAudiophile... but mostly, people here are too specialized and forget about that simpler aproach for regular people...

 

I'm a graphic designer for a newspaper in Montreal and if i was talking about InDesign, Photoshop or Illustrator to mansr or Jud (have nothing against you two (i'm sure you know), just for example...) you could be blown away from the perspective i'm talking about, for filters i used for tweaking images in Photoshop or text editing in InDesign, here, is the same, people talking about cables, filters, putting computer away from TV or wifi modem that changed sound... :-)

 

I don't know settings to upsampled best DSD for my DAC with Audirvana + or yours or should i do it in the first place... or should i used the plain settings coming from Damien's A+.

 

There is never an easy solution... some people like it simple... they don't change anything, some like it more difficult and some know what their doing... but here, NOBODY have the same settings or sound system or room distance .... that makes everything difficult to be simple  lol

 

Here, i learn something everyday...

 

My 2 cents for now...  Happy music everyone !!

[Jud]

16 hours ago, mansr said:

 

There's definitely room for more optimisations. Bear in mind, however, that much of the time is spent in the iZotope resampler, and that's none of my concern.

 

Yes, after you're done we can move on to bothering Alexey.

[Jud]

4 hours ago, RunHomeSlow said:

People like me (i think) really wants to have the best they could get EASILY from their software or DAC, looking in forums like here on ComputerAudiophile... but mostly, people here are too specialized and forget about that simpler aproach for regular people...

 

I'm a graphic designer for a newspaper in Montreal and if i was talking about InDesign, Photoshop or Illustrator to mansr or Jud (have nothing against you two (i'm sure you know), just for example...) you could be blown away from the perspective i'm talking about, for filters i used for tweaking images in Photoshop or text editing in InDesign, here, is the same, people talking about cables, filters, putting computer away from TV or wifi modem that changed sound... :-)

 

I don't know settings to upsampled best DSD for my DAC with Audirvana + or yours or should i do it in the first place... or should i used the plain settings coming from Damien's A+.

 

There is never an easy solution... some people like it simple... they don't change anything, some like it more difficult and some know what their doing... but here, NOBODY have the same settings or sound system or room distance .... that makes everything difficult to be simple  lol

 

Here, i learn something everyday...

 

My 2 cents for now...  Happy music everyone !!

 

Yes, very true.  How did you learn about filters in graphics and photography?

6 hours ago, Innocent Bystander said:

Interesting. 

Perhaps my perceptive is a bit different because my DAC does wonderful things, holographic soundstaging and remarkable clarity as fed with untweaked and un-upsampled A2.6.x, with a naturalness that makes me wary of change. I may  try v3 out of curiosity in due course, but I'm in no hurry, and in the meantime I.m hoping Damien will fix the library.

 

It is nice that A+ will accommodate both of our very different approaches.