Audirvana Plus 3 (official thread)

[Jud]

 

On my Mac Mini I can set 24bit depth in Audio Midi and also Dante/RednetI am thinking the top right readout in Audirvana has never shown 32/192 or 32/96, no matter what I set the various software to. So I think I'll A+ to limit to 24 bit incase the software tries to resample it.

 

My DAC is NOS R-2R and is 16bit 44.1 - 96K only.

 

You are confused. You don't have to bother to set A+ to 24 bits, but you can if you like - I'm guessing it will make no difference, but if you check it and it causes problems (unlikely), you can just uncheck it.

 

 

The 16 bit thing is about input, and you have a DAC that, as A+ shows, does 24-bit processing internally. That's fine. A+ will not "resample" based on the word length your DAC can handle internally. A+ will send your DAC what it can process as an input, which you have said is 16 bits.

[Jud]

still don't get why A+ always shows 32/something even when "Limit to 24" is set :-|

 

It's just telling you what your DAC is reporting as its internal processing (when it does internal processing, your DAC chip uses "32-bit float" math). Has nothing at all to do with what your DAC will accept as input, upsampling, etc.

 

The reason it's there in A+ is because a handful of DACs that do 24-bit internal processing don't behave correctly unless the player is told expressly they have this behavior. So if you were using A+ and you couldn't get any sound out of your DAC, that would be one thing you could try. But that's the extent of the setting's usefulness.

[Jud]

Those option are in place in case you have a DDC, like the Singxer F-1, between the computer and DAC that supports higher bit depths and sample rates than the DAC. Audirvana will not be able to determine the limits of the DAC and could try to pass bit depth and sample rates the DAC does not support.

 

I use a Singxer F-1. My DAC accepts a max input of 24/192 so I check the 24 bit option and to limit the max sample rate to 192 kHz.

 

Yes, useful in that case as well.

[Jud]

My DAC accepts a max input of 24/192 so I check the 24 bit option and to limit the max sample rate to 192 kHz.

 

If you connect your DAC directly, what word length does it show that it will accept?

[Jud]

I don't think the is correct. None of these bit depth number has anything to do with what bit depth math is done at. It is the bit depth and sample rate of the source material or what is being sent to the DAC. On the left side for the DAC input they are shown in this format:

 

24/176.4kHz

 

On the right, for source format, they are in this format:

 

16/44.1 kHz

 

The first number in both cases is bit depth sent to DAC or bit depth of source material, respectively. Not the number of bits the internal math is done at.

 

You're correct. Sorry for my confused thinking and thus misinformation. But it's still true that for the vast majority of cases the setting doesn't need to be done. Only where you have a digital-to-digital converter or your DAC doesn't report the input word length capability correctly might there be any problem at all. Only if you don't hear anything or you get some sort of distortion and think this might be the problem do you need to check the box.

[Jud]

I think V3 has some extra fields added to the plist (DB) so unless you take out that file and replace with the 2.6 version pre-upgrade, it will no longer open you albums in 2.6 or older versions. If using multiple A+ versions it is required to have the app support items backed up.

 

On sound quality, I think V3 is the same as 2.6, or very close (now not able to use 2.6 to A/B). Certainly not worse in any way I can tell. The bass on my system is still full and deep / huge. I run A+ with no EQ, no up sampling, 100% volume, Direct Mode, basically no messing with anything in the data path.

 

I am on a Mac Mini (tweaked) and it sounds great. I recommend the SSD and RAM maxed out, and the Uptone Audio 12V DC board with an external LPS. Also much prefer using AOIP over the USB out.

 

If you want to run both versions, I'm supposing you should probably keep separate database files. For my part I'm happy with 3.x, so 2.x is getting dumped.

[Jud]

I've upgraded from Audirvana 2.6 to 3.0.2. After the installation I cleared out my library as I wanted to start afresh. I then added a few folders to the Library tab of Preferences and they appear there but I am seeing nothing in my Library window, it is blank. If I toggle between Library and Favourites I see a brief flash of 'Loading' but Library remains empty. The windows for Qubuz and HiResAudio are functioning and files are playing as expected. How do I get the local folders/files to load to the Library Page? Is there something I need to to remove that's hanging over from the previous version, for instance. All help appreciated.

 

The first load of a library can take a long time to complete, and can take a while before you see any progress. I would suggest getting rid of the current database file, as it may have been corrupted if you closed the app thinking it was stuck, and starting over again. This time, start it off and then go do something else for a while.

[Jud]

Thanks Jud. Where I can I find the current database file? Is it a file in package contents?

 

 

Not home to look right now, but if I remember correctly the location is shown (and can be changed if you like) at the bottom of the Library preferences page.

[Jud]

A way to see what your settings are doing is to run the trial version of iZotope Rx.  It will show you a frequency response graph.  With some fiddling to make a pulse you can also get it to show you the results for ringing.

 

https://www.izotope.com/en/products/repair-and-edit/rx.html

[Jud]

@Sander_Kaa:

 

Go to User... Home Folder... Library... Applications Support... Audirvana Folder...

Trash the AudirvanaPlusDatabase.sqlite and the other files if you have more there.

 

Go to User... Home Folder... Library... Preferences...

Trash com.audirvana.Audirvana-Plus.plist

 

Restart MAC just to be sure and reopen A+ for new setup.

 

Edit: Sorry, this will not work for you.  See Damien's response below.

[Jud]

Whoops - @Sander_Kaa, Damien just gave the correct response.  So what I suggested won't work.  Sorry (you can deduct the reputation point you gave me  ).

[Jud]

28 minutes ago, Innocent Bystander said:

I don't upsample - I don't understand the point, as it can't create something non-existent.

 

The reason upsampling has existed since before there *were* (separate) DACs is because it makes the filtering necessary for digital to analog conversion easier to do at reasonable parts cost.  Your DAC almost certainly upsamples internally before the final conversion step from digital to analog.  So what pl_svn is saying is that the choice isn't whether or not to upsample, it's where: in your DAC or in software.  A lot of folks here like doing it in software, because we think it results in better quality.  Not trying to change the original performance/recording, trying to do a more accurate job of re-creating it.

[Jud]

7 minutes ago, TommyCat said:

I personally don't upsample and have no hi-res files past 96khz; I believe anything past that is both unhearable (is that a word?) and just puts stress on the system.  This is of course a highly arguable topic akin to the cable war (Yes, I do think cables make a huge difference ;-)

 

Since your DAC upsamples to 352.8 or 384kHz internally before sigma-delta modulating the signal to MHz rates, the lower resolution you feed, the more work your DAC must do.  Want to stress it less?  Feed it the highest resolution it will accept.

[Jud]

8 hours ago, TommyCat said:

If I am feeding my dac a 24/96 file, and the dac confirms that in the display (Benchmak HGC DAC2) why would it then upsample the file to 300+ kHz?.  Quite possible I am completely ignorant here.  I can send an email into Benchmark and maybe get my head screwed on straight.  For some reason I doubt this is happening since JohnSiau of Benchmark is quite passionate in his white papers against high up sampling rates.  

 

Maybe I have fallen trap to "it's not the things you don't know that hurt you, it's the things you know for sure that just ain't so"

 

Benchmark resamples everything to 220kHz in that DAC before delta sigma modulating to MHz rates, if I remember correctly.  I may be mistaken about the preliminary rate, but not about the fact it is a delta sigma DAC.

 

So a higher resolution is a little closer to the preliminary internal upsampling resolution of your DAC.  While this doesn't mean hi res will sound better, it shouldn't sound any worse unless the mastering was different, and it won't strain anything.

[Jud]

1 hour ago, astrostar59 said:

Interesting point made on up sampling. I am not in that camp, but bear with my thoughts on it. Wouldn't it be key to ensure any up sampling if performed in A+ be exact multiples of the data rate?

 

Only if you think computers can't do decimals.

 

Resampling to PCM rates is trivial for any computer.  Heck, the chip in your phone does it.

 

And no, the sample rate has absolutely nothing to do with "smearing" anything.

[Jud]

7 hours ago, OldBigEars said:

I use a Hegel HD25.  According to the LED display, it processes redbook files at 44 kbps.  I dunno, does that mean it upsamples them?     

 

According to the LED display, the *input* for RedBook files is 44.1 Khz - of course it is, that's the RedBook standard.

 

Looks like the AK4127 is a chip for doing asynchronous sample rate conversion (ASRC); my assumption is that everything you send the DAC is internally converted to 192kHz, judging by the AK4127 datasheet.

 

Edit: And is then sigma-delta modulated to MHz rates.

[Jud]

Both the Benchmark and Hegel are examples of DACs that do asynchronous sample rate conversion (ASRC) for jitter reduction purposes.  I can get into more of an explanation of that later.  ASRC has been mostly superseded by (similar-sounding but technologically different) asynchronous USB input.  The new(er) Benchmark actually does both.  Their older DAC just did ASRC.

[Jud]

10 hours ago, TommyCat said:

Well, that is beautiful and thank you!  Looks like upsampling to 211 khz is happening!  So the theory is that upsampling in the software, prior to sending to Dac, is more efficient than letting the dac do it?

 

thanks again!

 

10 hours ago, wwaldmanfan said:

I don't believe that the DAC has to "work harder" to do what it is designed and built to do. By the same token, upsampling in the music player software forces the computer to "work harder" before sending the file to the DAC, does it not?

You have options, so assume nothing. Better to experiment a bit, and let your ears decide what sounds best. On my rig, I think that iZotope upsampling in Audirvana colors and muddies the sound, and artificially alters the soundstage. YMMV.

 

This does not have to do with "greater efficiency" in the way that you are thinking of it.  The filtering on a DAC chip isn't difficult for the chip and is trivial for a computer.

 

What upsampling in software is about is, first, the ability to run *better* (to some extent, more sophisticated) filtering on a computer than can be programmed into a DAC chip.  Compared to the CPU in a computer, a DAC chip has more limited resources to run sophisticated filtering.  FPGAs have more resources for this purpose than DAC chips (though still not quite what a computer CPU has), and if you look at DACs that use FPGAs for this purpose you will find many of them among the most expensive DACs available (though Chord has notably brought DACs with FPGAs well down in price point).

 

Now Chord has a very specific philosophy about what kind of filtering it does in its DACs, which brings up the second point about software upsampling: It enables you to have *choice* in the type of filtering you do or the parameters you set for the filtering.  So you don't have to be bound to the particular choices made by the designer at Chord or the designer of your DAC or the designer of the firmware-programmed filtering in the chip inside your DAC.  Some people don't care to fuss with changing, which is very understandable.  Of course with Audirvana Plus if you don't want to play around with different filter settings, you have the choice not to upsample, or if you do upsample, to leave all the filter parameters at their default settings.

[Jud]

11 hours ago, fbczar said:

Which upsampling type/rate/multiple do you prefer in Audirvana for delta sigma DACs?

 

I upsample everything to DSD256.  Whether you are able to do this varies by DAC, of course.

[Jud]

1 hour ago, astrostar59 said:

I am also wondering if A+ is used to upsample, but that resolution was not the exact same value as the end DAC up samples to, are we not having the situation where the (possible) degradation will happen twice. If however, we send data to the end DACs sample rate upper limit, then would that 'shut down' the resampling function of that DAC. Unless it was disabled by doing this, in my mind we have 2 stage resampling / processing of the data. 

 

And what you are saying, all this is to negate the effect of the digital filter?

 

Getting more into the internals of most DACs:

 

- If you send it a RedBook file, it does 3 rounds of doubling the sample rate, first to 88.2kHz, then 176.4kHz, then 352.8kHz

 

- If you send it a file based on a 48kHz rate, it doubles until it reaches 384kHz, so a 96kHz input would go through two rounds of doubling

 

- From there (still on the DAC chip) the bitstream goes to sigma-delta modulation, where the sample rate is increased to a minimum of 2.8MHz, very often in newer DACs higher than that (in ESS DAC chips, I think it may wind up higher than 40MHz before all is done)

 

So if you have a 44.1kHz file and you choose to upsample in A+ to the highest rate the DAC accepts or the nearest 2x rate (let's say 176.4 or 192kHz for a DAC that doesn't take DSD input), you've avoided a couple rounds of doubling.  But more important than that, you have the opportunity, for the initial and most critical stage of upsampling, to employ different and possibly "better" filtering (for some value of "better," whether it be particular measurements, your listening enjoyment, or both) in software than the filters programmed into the firmware in your DAC chip.

[Jud]

24 minutes ago, audiocanyon said:

Since I have the Benchmark DAC2 also, I'm inspired from these various posts to do some A/B comparisons between the A+ upsampling and John Siau's.  Gotta love audio tweaking...

Sierra Direct Mode.pdf

 

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.

[Jud]

25 minutes ago, progfan2 said:

Hi Jud,

 

I running 2.6 have found using any DSD mode on my 2011 I5 Mini, stresses the CPU with heat running about 95C.  Would you know if A+ 3 using DSD mode has the same CPU demands.

 

Thanks,

Mark

 

I'm unaware of anything that would change the CPU usage, but Damien and mansr would obviously know better than I do.

[Jud]

19 minutes ago, mansr said:

 

I haven't done anything to reduce CPU usage. Perhaps I should.

 

It depends.  If reducing usage = greater efficiency, wonderful.  If it = slowing down the conversion process noticeably, obviously not as wonderful.

 

I'm guessing your code was already pretty efficient, so what would you contemplate changing?  Taking advantage of more modern CPU capabilities where they exist...?

[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."