Audirvana Plus 3 (official thread)

[pl_svn]

16 minutes ago, TommyCat said:

why would it then upsample the file to 300+ kHz?

 

 because all sigma-delta chips do, so unless you know for sure (sorry, too much stuff to read on its webpages: couldn't find it myself) it's a pure NOS DAC using instead R2R chips or some FPGA running a secret sauce... yours does too  

 

what you see on its display is just what's coming in

[jhwalker]

17 minutes 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"

 

FWIW, here is a good summary of the technology behind your DAC, with answers to these questions direct from John Siau:

 

http://www.audiostream.com/content/benchmark-media-systems-dac2-hgc-dac#VyZFs8K05WgpItRP.97

[TommyCat]

5 minutes ago, jhwalker said:

 

FWIW, here is a good summary of the technology behind your DAC, with answers to these questions direct from John Siau:

 

http://www.audiostream.com/content/benchmark-media-systems-dac2-hgc-dac#VyZFs8K05WgpItRP.97

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!

[wwaldmanfan]

1 minute 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!

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.

[OldBigEars]

1 hour 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 ;-)

TommyCat - agreed on all points.  I went through an extended loop - testing all kinds of upsampling to high res PCM and DSD, as well as fiddling around with  filtration tweakery.  Eventually I came to the realization that non-upsampling sounds best on my system.  It's like drinking coffee without lumps of sugar and dollops of milk distorting the flavor.  I just like it better. And yes, good cable selection makes a far more satisfying contribution to enjoying the music. YMMV :-)

[wgscott]

19 minutes ago, OldBigEars said:

TommyCat - agreed on all points.  I went through an extended loop - testing all kinds of upsampling to high res PCM and DSD, as well as fiddling around with  filtration tweakery.  Eventually I came to the realization that non-upsampling sounds best on my system.  It's like drinking coffee without lumps of sugar and dollops of milk distorting the flavor.  I just like it better. And yes, good cable selection makes a far more satisfying contribution to enjoying the music. YMMV :-)

 

I forget.  Which non-upsampling DAC are you using?

[TommyCat]

5 minutes ago, wgscott said:

 

I forget.  Which non-upsampling DAC are you using?

Thanks to jhwalker discovered my dac does upsample to 211 kHz (Benchmark HGC dac2).  Question remains whether or not to upsample before sending to dac (my tendency would be not to; assume Benchmark has figured out to do that more efficiently than my guesswork inside of A+)

[OldBigEars]

1 hour ago, wgscott said:

 

I forget.  Which non-upsampling DAC are you using?

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?     

[audiocanyon]

13 hours ago, trybeingarun said:

I am not able to get Interger mode to work properly. I am using Sierra 10.12.3 and have replaced IOAudioFamily.kext to get Direct Mode to work. Everything seems to work fine when with integer mode turned on until i plug the USB cable out or there is a power failure and the dac dies. When that happens my mac crashes. Anybody else experiencing the same issue?

 

The replacement IOOAudioFamily.kext with direct mode and either integer mode 1 or 2 works fine for me with Sierra.  I'm using a Benchmark DAC2.

[TommyCat]

5 minutes ago, audiocanyon said:

 

The replacement IOOAudioFamily.kext with direct mode and either integer mode 1 or 2 works fine for me with Sierra.  I'm using a Benchmark DAC2.

Can you give some details on how you did that?  Would love to be able to use Direct mode.

[wgscott]

15 minutes 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?     

This is all I found:  http://www.avmentor.net/reviews/2014/hegel_hd25_1.shtml

Quote

Main digital processing in the HD25 is performed via three Asahi Kasei chips. The trio includes the AK4115 digital interface, the AK4127 sample rate converter and -finally- the AK4399 d/a converter, which operates at 32bit. Of these three components, the role of the SRC seems to need some clarification. Hegel does not mention that the converter performs any kind of upsampling (all AK DAC chips can handle all standard sample rates, anyway) so the usefulness of a sample rate converter at this point could be associated with the re-clocking the company seems to carry out, but without giving more details, one cannot say for sure. 

Someone who knows a lot more about this stuff than I do would have to help us out here. Sorry.

[bhobba]

Damion can you verify if 3,1 will work with the Direct Stream Bridge 2?

 

Currently I just use Bubble UpNp into my Direct Stream Bridge 2 to play Tidal.

 

Will the 3.1 version allow me to do the same with Audirvana.  

 

The sound quality via Bubble UpNp is quite good, but Audirvana using a good USB cable easily bests it - more alive and dynamic - Bubble UpNp is sort of dull and thick by comparison.

 

I know the bridge 2 has a software update coming to do MQA, but I still suspect what you will do is likely better.

 

This is a big decision for me.   I will need to a new Mac.  If its coming will get a new Mac and Audirvana 3 straight away.   Already have a copy of Audirvana 1 but since then my Mac bit the dust.

 

Thanks

Bill

 

 

 

 

[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.

[astrostar59]

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? Isn't that line of the points made by the MQA team? I may be wrong, but I thought going outside of that smears / degrees the sound, as basically it is resetting all the data point and guessing the sound waves all over again, bit like the loss from the studio master higher resolution files going down to Redbook, and yet more loss trying to take it back up again?

 

Also, isn't it also important to find out what the relevant DAC is doing. It that DAC is again up sampling in the chip, we have yet another process to smear the data. Obviously testing and listening will help, but from my understanding it seems counter productive.

 

I understand all this is in theory to control the digital filtering, is that correct? 

 

If we are changing the format from Redbook PCM to DSD to then use a FPGA then that may make sense as that has a different topology and 'may' improve things?

 

Sorry, I am old school R-2R and hit 96K limit on my DAC. I must say, I have heard a few DACs without a digital filter, and they do seem (to me) to sound better, and my speakers haven't blown up or my ears caught on fire. Only saying....

[trybeingarun]

audiocanyon, do you mind sharing your file? May be something's wrong with my file. Are you using latest Sierra version (10.12.3)? I would love to use integer mode, but I can't risk my mac crashing everyt ime I use integer mode.

[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.

[Giacomino]

1 hour ago, trybeingarun said:

audiocanyon, do you mind sharing your file? May be something's wrong with my file. Are you using latest Sierra version (10.12.3)? I would love to use integer mode, but I can't risk my mac crashing everyt ime I use integer mode.

https://audirvana.com/forums/viewtopic.php?f=5&t=1048

 

https://dl.dropboxusercontent.com/u/19274252/AudirvanaSierraDirectMode.zip

[mevdinc]

12 minutes ago, Giacomino said:

https://audirvana.com/forums/viewtopic.php?f=5&t=1048

 

https://dl.dropboxusercontent.com/u/19274252/AudirvanaSierraDirectMode.zip

The above Dropbox link is dead, the working link is below.

Re: Direct-Mode in macOS Sierra

• Quote

Post by RunHomeSlow » Thu Mar 16, 2017 11:42 pm

Since yesterday, Dropbox changed his security, so my link is dead. Here it is for those who still wants it...
https://www.dropbox.com/s/ex3ksuj1calu7 ... e.zip?dl=0

[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.

[astrostar59]

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?

[audiocanyon]

10 hours ago, TommyCat said:

Can you give some details on how you did that?  Would love to be able to use Direct mode.

 

I followed the attached instructions.  I can't remember who provided these (Katsu?), but I picked them up off the A+ 2.6 forum and they worked perfect for me.  It sounds like you have the El Capitan IOAudioFamily.kext file, but if you don't let me know.

Sierra Direct Mode.pdf

[27Globes]

Trying out Audirvana 3.0... but find Library Manager confusing.. trying figure out how to filter out library by genre. Looked over user manual... still don't get it.  Can someone layout the procedure... would appreciate.  Does not seem as accessible as it was when integrated with iTunes.  Thanks.