Audirvana Plus 2.5.1 with PCM->DSD Conversion

[Outlaw]

But I don't feel like building twice the computer.

[huaklom]

That's because it's twice as good ;-)

 

For SQ we will see some day soon, but for easy navigation thru iPad, I give you x3 or x4, not twice.

[jhwalker]

DSD works ok.But it uses twice as much processor power then hqplayer.

 

I'm not seeing that. On my computer, HQPlayer typically consumes about 250% CPU (DSD128, poly-sinc, ASDM7), while Audirvana upsampling to DSD128 uses about 220-230%. So HQPlayer is actually more CPU-intensive. And Audirvana never "stutters", like HQPlayer does sometimes - if overloaded, it simply pauses and waits for the resampler to catch up.

 

I'm glad we have two really good options

[OldBigEars]

I was really enjoying the B3 filter, through my MBP with 8GB memory/ 480 SSD and Concero HD DAC. Absolutely no glitches until after about an hour, I suddenly lost sound through the left channel. I was playing a FLAC 16/44 file. My Isotope settings:

- 83

- 800,000

- 1.08

- 200

- .30

 

I'm a bit concerned about the overheating issue. I don't know how to determine how hot my computer is actually running but the corner around my cord connection got quite warm. I've decided to give DSD a break for now. Look forward to hearing more experiences from others

[Jud]

Steepness: 23

Filter max length: 1,900,000

Cutoff freq: 1.02

Anti-aliasing: 200.0

Pre-ringing: 0.55

 

I'm certainly open to making some changes if it addresses the PCM->DSD issue.

 

--David

 

I have no idea if it will do anything, but since it's easy enough to try, see what happens with cutoff and pre-ringing both set to 1.

 

Edit: And you might try max length 10,000.

[Jud]

Something interesting:

 

Tried converting a few files to DSD128 with HQPlayer and A+, and monitoring CPU usage with top.

 

With 16/44.1 files, A+ CPU load was between 174 and 190+%. With HQP it was mostly around 110% with an occasional spike to 150 or 175%. No stuttering or distortion with either.

 

With 24/192 files, exactly the same result with A+. With HQPlayer, CPU% was now in the 180-190+% range, and the one track I tried stuttered continously and was thus unplayable.

[qdwieertteyrujpo]

I really do enjoy the sound after converting to DSD. Unfortunately, my MBP fan (mid 2012) does not seem to be a "fan" and protests rather loudly a few minutes into play:-)

[TheDuke]

Hi all, I've been a long time reader of this forum and and user of Audirvana since 2012. Greetings from South Wales UK.

I'm using an IFI IDAC2 with the latest firmware 5.1. I briefly used Audirvana Plus 2.5.0.6 and lost the left channel intermittently, on first playback. I have now started using 2.5.0.7 and all is working well, the sound is great, and upsampling to DSD128 without issue. The IDAC2, I believe is capable of DSD256 ,but this is greyed out in the Active Audio Device info. Does anyone know how enable DSD256.

[Dyson]

It would be helpful if you could elaborate on that. Also, what DAC do you use?

 

Sure I want to elaborate, just needed some more listening.

 

DAC: iFi micro iDSD v5.1

iZotope: preset max

upsampling: DSD256

source files: 44.1/24

 

Type A filters: they sound fine nothing really to mention, a kind of "all-round". I can't really tell much differences between the versions (maybe A0 and A4 the best?).

 

Type B filters: they are really very organic, full of body and rounded (attacks are a bit slow). Version 0 sounds the best for me, the other versions are missing a bit of depth (B2 being the best after B0).

 

Type C filter: keeps the full body and depth of the B0 filter and adds more attack (more sparkling).

 

 

I have as well a track that as a clipping issue for all filter: Barb Jungr - Many Rivers to Cross from Linn (192/24). When she starts to sing at 4.5 sec. the volume is suddenly decreased/cut.

[Miska]

I'm not seeing that. On my computer, HQPlayer typically consumes about 250% CPU (DSD128, poly-sinc, ASDM7), while Audirvana upsampling to DSD128 uses about 220-230%. So HQPlayer is actually more CPU-intensive. And Audirvana never "stutters", like HQPlayer does sometimes - if overloaded, it simply pauses and waits for the resampler to catch up.

 

I don't think one can easily compare load of the two, at least with HQPlayer the load varies quite a lot depending on material and such because of the algorithms. At least the modulator mansr has in SoX is much simpler/straightforward so it will likely consume fairly constant amount of time per input sample.

[Dyson]

Just a follow up:

 

Testing now with iFi Micro iDSD, direct from MacBook Pro (i.e., no reclocker or split cable, just direct from the MacBook to DAC), the glitch at 5 seconds is still consistent for all music at multiples of 44 sample rate. The failure to playback (i.e., no playback, stuck on "Loading first track . . . ") is, if anything, worse (or, let's say, "more consistent"): with the iFi, I have to pull out and reset the USB cable with each new playback session. That is, it will play back multiple tracks consecutively, but if I click "stop" to move on to another selection, playback will fail every time until I pull out / plug back in the USB cable, at which point, it will play fine until the next "stop".

 

So I'd think this has something to do with re-initializing the USB connection after "stop" - just speculating.

 

I have also a glitch with my micro iDSD at 5 seconds, but only when doing DSD128 conversion (not happening with DSD64 and DSD256) AND only with a 96/24 track. The same 44.1/16 track plays fine. (only tested type C filter)

[Le Concombre Masqué]

kudos.

 

Might end up upgrading my 1.5 license after a long pleasant time with HQP...

 

better soundstage, air, "thereness" at a live event feeling ; clean detailed uncluttered smoothness

 

B3 was a good suggestion. I settled temporarily with B4 but will recheck Dyson's favorites.

 

Does playing with Isotope advanced settings bring anything good to someone's ears ? Then, recommended settings please ?

 

 

Wish DSD 64 was up sampled to 128

 

0 issue whatsoever with a TEAC 501

 

Testers, don't forget to adjust volumes ! thing is we all lower volumes with HQP to avoid "Limited" to increment ; hope A+ doesn't loose its SQ in the problems solving process. Always preferred HQP at 0 Vol but settled to -3 just not to bother...

[Jud]

I don't think one can easily compare load of the two, at least with HQPlayer the load varies quite a lot depending on material and such because of the algorithms. At least the modulator mansr has in SoX is much simpler/straightforward so it will likely consume fairly constant amount of time per input sample.

 

That appears to match what I saw.

 

The interesting/puzzling thing for me is that I got stuttering with HQPlayer at a CPU load about equal to a load where I did not get stuttering with A+. Is the load an average and stuttering occurs at momentary higher loads? Or is there another explanation?

[Jud]

I really do enjoy the sound after converting to DSD. Unfortunately, my MBP fan (mid 2012) does not seem to be a "fan" and protests rather loudly a few minutes into play:-)

 

Some info on DAC, iZotope settings, SDM "filter," etc.?

[mansr]

I was really enjoying the B3 filter, through my MBP with 8GB memory/ 480 SSD and Concero HD DAC. Absolutely no glitches until after about an hour, I suddenly lost sound through the left channel. I was playing a FLAC 16/44 file.

 

Does this problem happen consistently with that file?

[mansr]

Hi all, I've been a long time reader of this forum and and user of Audirvana since 2012. Greetings from South Wales UK.

I'm using an IFI IDAC2 with the latest firmware 5.1. I briefly used Audirvana Plus 2.5.0.6 and lost the left channel intermittently, on first playback. I have now started using 2.5.0.7 and all is working well, the sound is great, and upsampling to DSD128 without issue. The IDAC2, I believe is capable of DSD256 ,but this is greyed out in the Active Audio Device info. Does anyone know how enable DSD256.

 

If you're using DoP, you need the 5.1A firmware for DSD256.

[OldBigEars]

Does this problem happen consistently with that file?

 

I've played this file (happens to be Corinne Bailey Rae "Seasons Change") with B4 with no issues. I was enjoying B4 last night - at the moment it's my favorite. Big, round, nuanced sound. But my CPU consistently runs 250-260% - which concerns me. Should it?....I'll probably try some DSD64 today - maybe this is the sweet spot with less overheating.

[damien78]

i've been playing with 2.5.0.7 this morning and greatly enjoying the sound, when it works

 

all tests performed at dsd128, have not tried dsd64 much, just to confirm it works.

 

2011 macbook pro i7, 2.5ghz, 16gb ram, 1tb ssd. Lh labs lightspeed 2g usb cable, w4s recovery, lh labs geek pulse x infinity (powered by lh labs lps), alpha prime.

 

Observations:

• dsd > dsd resampling does not work, just plays the original rate dsd, probably by design
  

That's correct, the upsampling is only for a PCM material

 

• au processing is not possible, aus are simply ignored - is it possible to do the processing first to the pcm stream, then convert to dsd for output, so we can get the benefit of, for example, equalization and / or crossfeed?

I still have to implement this, though the processing will be only during the loading phase. The realtime effects (AU & dithered volume control) are not active when playing DSD material (upsampled or native).

 

• in general, the first 0.5 - 1 second of each track is cut off if you start playback on that track - not necessarily noticeable if there is a silence at the beginning, but consistent if the music starts at the beginning of the track (e.g., classical music intended to be gapless)

This means your DAC takes time to settle on DoP mode. You can set a sample rate change additional latency of 1 or 2 second to fix this. (Option in the Audio System page of the preferences)

 

• gapless works well; i.e., there is no cutoff of the first 0.5 -1 second of the track if proceeding from one track to another
  
• at all sample rates which are multiples of 44, there is a playback glitch / gap at 5 seconds into the playback. Very consistent with 44, 88, 176, 352 sample rates, and it always happens at 5 seconds, independent of the sample rate
  
• sample rate that are multiples of 48 do not glitch at 5 seconds
  

Can you give your iZotope filter settings?

And how would you characterize this glitch at 5 seconds for the first playing track? (duration, noise/silence,... ?)

 

• at least half of the time, playback hangs loading the first track. When this happens, cpu utilization is near 0; i.e., nothing is happening
  
• the playback hangs seem to be indiscriminate to the filter type, with a few caveats: A0, b0, and c seem to hang most frequently, but i've gotten them to play at least a few times - so there are no filters that simply don't work, ever. A2 and b1 seem to work every time.
  

The hang on "Loading first track..." may also well be due to an initialization issue of the DAC.

Resetting the DAC USB interface (or driver on the Mac) by unplugging then replugging the USB cable fixes it as per your experiment.

But when this happens, how long have you let between the previous stop (I mean full playback stop, not pause), and the request to play the new track?

[damien78]

I'm still experiencing the same symptoms with 2.5.0.7 as I described with 2.5.0.6. With 2.5.0.7, I tried Filter Types A0, A4, B0, B4, and C. None of these filters made a difference with the noise issues.

 

I did notice that when attempting to convert 24/192 PCM to DSD64, the spurious tones seemed to appear at exactly the same points in the track when trying different filters.

 

This time around, I also tried playing some DSF DSD64 tracks. Non-upsampled DSD64 plays without issue (do the different filters work in this case?). If I try to upsample DSD64 to DSD128, playback is fine, but there's no upsampling. This is shown by both A+ and the display on the Mytek.

 

When doing the PCM->DSD test, A+ would crash about every two filter changes, at the point when I tried to start playback after changing the settings.

 

My PCM test track is Track 1 ("Natsukashii") from Natsukashii by the Helge Lien Trio (24/192). The DSF test track is Track 1 ("These Are Soulful Days") from Lee-Way by Lee Morgan (2008 remaster, DSD64). DAC is Mytek Stereo192-DSD, latest firmware, latest FireWire drivers.

 

--David

 

What is the DSD filter setting on your Mytek? 50, 60 or 70kHz LPF ?

 

What do you mean by "A+ would crash about every two filter changes"? Is it an application crash (in this case, can you send me by email or PM the crash report)?

 

Damien

[Foggie]

Damien nice work. I was shocked to see this thread as I wasn't expecting to see DSD conversion functionality in A+. I assume the latest remote app isn't built to work with these beta's?

 

I took a quick spin and all seems to be working and sounds good. As others have noted, the PCM-DSD conversion is very CPU intensive and my mini (2.3Ghz i5) sits right around 220%-235% the entire song (SSD, 8GB mem). Native DSD files are the opposite (obviously) => ~30% on initial load which loads to mem pretty quick then settles down.

 

In comparison, HQP with NAA on a Win wkst (W3680, SSD) hovers around 6-10%. Not apples to apples (hrdwr), just showing my results in my environment.

[AnotherSpin]

Didn't expect I will pay any serious attention to A+ anytime soon - DSD conversion provided by HQP was best solution within OS environment I knew. Still I was curious enough to check this thread and after I realized A+ is doing DSD conversion now I decided to hear the results myself. Downloaded trial (I never purchased A+2 license) and started my listening. Well, it is very impressive! As per now I couldn't agree with some colleagues who started celebrate the end of HQP already, however, I should admit DSD 256 with A+ gives an overall sound impression at least on par with HQP, and better in certain respects (mac mini, ifi micro idsd).

[Jud]

That's correct, the upsampling is only for a PCM material

 

• 
  

I still have to implement this, though the processing will be only during the loading phase. The realtime effects (AU & dithered volume control) are not active when playing DSD material (upsampled or native).

 

• 
  

This means your DAC takes time to settle on DoP mode. You can set a sample rate change additional latency of 1 or 2 second to fix this. (Option in the Audio System page of the preferences)

Can you give your iZotope filter settings?

And how would you characterize this glitch at 5 seconds for the first playing track? (duration, noise/silence,... ?)

The hang on "Loading first track..." may also well be due to an initialization issue of the DAC.

Resetting the DAC USB interface (or driver on the Mac) by unplugging then replugging the USB cable fixes it as per your experiment.

But when this happens, how long have you let between the previous stop (I mean full playback stop, not pause), and the request to play the new track?

 

With .7, when double clicking the first track, I get the "Loading first track..." hang exactly every other time (in other words, the 1st, 3rd, 5th, etc. attempts). But the 2nd, 4th, etc. attempts, it starts up just fine, no other fiddling necessary (no plugging or unplugging or anything else).

[damien78]

With .7, when double clicking the first track, I get the "Loading first track..." hang exactly every other time (in other words, the 1st, 3rd, 5th, etc. attempts). But the 2nd, 4th, etc. attempts, it starts up just fine, no other fiddling necessary (no plugging or unplugging or anything else).

 

What if you wait about 5 sec. ? Is it still hung?

[Jud]

What if you wait about 5 sec. ? Is it still hung?

 

I have waited what seemed to me some while, but as I am not very patient, I may not have made it to 5 seconds. I will see when I am home again.

[orgel]

I have no idea if it will do anything, but since it's easy enough to try, see what happens with cutoff and pre-ringing both set to 1.

 

Edit: And you might try max length 10,000.

 

So this is interesting: If I change the three parameters as suggested, no more noise issues. A+ just crashes immediately when I hit Play. If I change back to my previous settings, it's back to the same old noise problems.

 

--David