Audirvana 1.5 Beta

[damien78]

Hi Damien

Thanks you, this release works with the IK Multimedia ARC 2 & Fabfilter.

But with the TC Electronic AU " Assimilator Native, Integrator, M40 Reverb, ResFilter Native" nothing appear in the list.

 

With Apple Logic Pro, this AU passed and are displaying...

 

Have a nice day !

Are those plugins available only in 32bit ? In this case you can try to launch A+ in 32bit mode to check. ... and inquire with TC Electronic about a 64bit release.

 

Bonne journée,

Damien

[Encore]

Hi Damien,

 

As I wrote earlier, SQ once again got better. However, I have a problem, which I also had with 1.4.6: After I upgraded from Snow Leopard to Mountain Lion, the player regularly just stops. It doesn't crash--the music just stops and A+ "forgets" where it got to in the playlist. No need to restart A+, but the music must be restarted. Any ideas what is happening?

 

I have a MacbookPro i5 w 8 GB ram and 320 GB hd.

[damien78]

@Encore: Can you send me in PM the Debug Info ?

 

Damien

[brlawyer]

I agree with you and Jud and also applaud hard Damien to his incredible effort to please every one, even with the plug-in gadget. Regarding iZotope it's a very different matter than the "Audio Units", but as I said in my first reply, you have the choice to use it, or not. I guess @brlawyer is in my same way of thinking, if not, we disagree.

 

Roch

 

Yes; we are on the same page, Roch.

 

I don't mind if the option is there for anyone to use it; but I just find it odd that, after years of preaching the opposite, we see a new "equalizer" in A+ (or other audiophile players for that matter). From Damien's own white paper:

 

"The computer is a great music server but also asource of jitter and other RF interferences thatare detrimental the sound quality, even whenbit-perfect reproduction is ensured.

 

The player software needs to optimize andstreamline the audio path to minimize theseadverse effects essentially linked to theprocessing load synchronous to the audiostreaming. Achieving “source direct” in additionto “bit-perfect” is key. "

 

 

 

 

You can obviously conclude from the above that the main if not only purpose of an audiophile-grade player like A+ has always been to ensure the most unadulterated reproduction of sound possible, i.e., "bit-perfect+source-direct".

 

Unless Damien can explain further, the use of AUnits defeats that principle completely, as it adds to the "User Space" the same distortions that "usual audio players" have when using Core Audio units.

 

Sorry for being too dogmatic here - it's just my lawyer's "consistent logic" mind working again.

 

p.s: On a totally unrelated note, does anyone know if my Nuforce Icon HDP is USB-asynchronous? I cannot find this information anywhere.

[Cycleman]

Sorry for being too dogmatic here - it's just my lawyer's "consistent logic" mind working again.

 

But music is not logic. Music is emotions.

 

So perhaps "consistent logic" is just the wrong metric here.

 

What many of us are trying to create, and in our subjective ways measure, is pleasure and not necessarily exactness.

 

If all the audio toys and tools make me (make you) feel good, perhaps that's the principle.

 

(or at least that's a whole 'nother way to look at it)

 

Dave, who sent out this quote to his list just the other day:

Music is love in search of a word.

--Sidney Lanier

[brlawyer]

But music is not logic. Music is emotions.

 

So perhaps "consistent logic" is just the wrong metric here.

 

What many of us are trying to create, and in our subjective ways measure, is pleasure and not necessarily exactness.

 

If all the audio toys and tools make me (make you) feel good, perhaps that's the principle.

 

(or at least that's a whole 'nother way to look at it)

 

Dave, who sent out this quote to his list just the other day:

Music is love in search of a word.

--Sidney Lanier

 

Sure; let me just summarize what I said above - my beef is with the "preaching one approach now and another later", that's all.

 

Of course what ultimately matters is the pleasure you get from the listening experience - let's just try to be more consistent when advertising certain "bit-perfect advantages" next time.

[Encore]

There's no debug info as far as I know. The music just stops--the program keeps running.

[orgel]

There's no debug info as far as I know. The music just stops--the program keeps running.

 

Next time the problem recurs, select "Debug Info..." from the Audirvana Plus menu.

 

--David

[edwardsean]

Sure; let me just summarize what I said above - my beef is with the "preaching one approach now and another later", that's all.

 

Of course what ultimately matters is the pleasure you get from the listening experience - let's just try to be more consistent when advertising certain "bit-perfect advantages" next time.

 

Okay, I tried so hard to stay out of this debate, but I can't seem to help myself. So just one post and I'll stop there.

 

Music is both logic and it's not, absolutely. More precisely then, we are talking about music reproduction, and there is no inconsistency in Damien's approach. The goal is to build as pure a signal pathway as possible, so as to be rid of all uncontrolled sonic distortions, artifacts, and anomalies. That's what he's doing. From there, the ability to add plugins is to offer the ability to add--controlled, manipulable--sonic distortions or anomalies.

 

Those who want to add EQ, time, phase, harmonic enhancement still want as clean a signal base as possible. Audio purists may not want this, and that's fine, it's an additional feature. But then again, audio purists may want this to correct the EQ, phase, time distortions and anomalies in their listening environment. Either way the goal of providing fundamental signal purity is the same and remains unchanged. We all want the same thing here and hats off to Damien for getting us closer and closer.

[Melvin]

Okay, I tried so hard to stay out of this debate, but I can't seem to help myself. So just one post and I'll stop there.

 

Music is both logic and it's not, absolutely. More precisely then, we are talking about music reproduction, and there is no inconsistency in Damien's approach. The goal is to build as pure a signal pathway as possible, so as to be rid of all uncontrolled sonic distortions, artifacts, and anomalies. That's what he's doing. From there, the ability to add plugins is to offer the ability to add--controlled, manipulable--sonic distortions or anomalies.

 

Those who want to add EQ, time, phase, harmonic enhancement still want as clean a signal base as possible. Audio purists may not want this, and that's fine, it's an additional feature. But then again, audio purists may want this to correct the EQ, phase, time distortions and anomalies in their listening environment. Either way the goal of providing fundamental signal purity is the same and remains unchanged. We all want the same thing here and hats off to Damien for getting us closer and closer.

 

Very well said, and at the same time I understand completely where brlawyer is coming from. For years we obsessed over bit-perfect and now it seems no one actually gives a sh$t. Up-sampling has also been debated to death on this forum for the same basic reasons. Audio Units is simply the latest.

[Superdad]

Very well said, and at the same time I understand completely where brlawyer is coming from. For years we obsessed over bit-perfect and now it seems no one actually gives a sh$t. Up-sampling has also been debated to death on this forum for the same basic reasons. Audio Units is simply the latest.

 

While I agreed with edwardsean, I don't agree with your statements Melvin:

a) I think we do still care, not just about "bit-perfect," but also about having software that we trust to bypass the layers of the OS as much as possible and speak as directly as possible to the DAC, while eliminating extraneous processing and computer processes. That includes Hog/Exclusive modes, Integer Mode, memory play, and whatever even deeper file segment fetching/paging/caching optimization that Damien has been doing these past months that have resulted in large leaps in SQ.

 

b) If I feel inclined to try my hand with digital room correction (I already have a measurement mic and USB mic-pre/A-D converter; I just need to teach myself one of the s/w packages), the only way I would want to apply it (a resulting correction file) would be via a plug-in to A+. Otherwise, as some have said, all the other A+ benefits go out the window.

 

c) In a way, upsampling is a good example, as the iZoptope SRC engine that Damien licensed is, in effect, a hard-coded plug-in to A+. Moreover, since just about every DAC on the planet (save for true NOS/filterless R2R ladder DACs) has a digital filter of some sort (good, bad, or ugly) that we are forced to listen to (whether the DAC upsamples or not), the beauty of iZotope in A+ is that we are given the power to design our own filter, and the higher the rate we upsample to, the more we push our DACs' built in filters out of the way (See my post #116 in this thread for a recent experience). While a few finished DAC manufacturers design their own discrete filters (i.e. as opposed to using the ones built into the DAC chip), most just pick one of the built in choices (or give us a choice). A sigma-delta DAC chip by its nature must oversample (8x from 44.1), so there is no getting around this. But we can prefer to create a stream that is closer to the DACs internal "native" rate using customizable s/w.

I look forward to the day when more USB DACs accept 352.8/384; A+ stands ready to feed them.

 

d) If the CD standard--and thus the majority of humanity's digitally released music--had originally been set at 24/96, then we likely would not be having any discussions about up/oversampling and digital filters. We would all happily be listening to filterless NOS DACs. But as the owner of an NOS DAC, I can not go back to listening to straight 16/44.1, it is just too "dirty."

[Jud]

I don't mind if the option is there for anyone to use it; but I just find it odd that, after years of preaching the opposite, we see a new "equalizer" in A+ (or other audiophile players for that matter). From Damien's own white paper:

 

"The computer is a great music server but also asource of jitter and other RF interferences thatare detrimental the sound quality, even whenbit-perfect reproduction is ensured.

 

The player software needs to optimize andstreamline the audio path to minimize theseadverse effects essentially linked to theprocessing load synchronous to the audiostreaming. Achieving “source direct” in additionto “bit-perfect” is key. "

 

You can obviously conclude from the above that the main if not only purpose of an audiophile-grade player like A+ has always been to ensure the most unadulterated reproduction of sound possible, i.e., "bit-perfect+source-direct".

 

Unless Damien can explain further, the use of AUnits defeats that principle completely, as it adds to the "User Space" the same distortions that "usual audio players" have when using Core Audio units.

 

Sorry for being too dogmatic here - it's just my lawyer's "consistent logic" mind working again.

 

 

When you find your "logic" becoming more dogmatic than the developer's thinking, it should make you stop and think, and it has. Good. So now why does your logic reach a different conclusion than Damien's? Likely you are beginning from different premises. Now whose premises are more likely to be missing key information? Right, exactly. :-)

 

I'm not Damien, but let me see if I can put forward some thoughts that may help.

 

What you seem to be looking for - me too, believe it or not - is simplicity in service of fidelity. So let's look at "bit perfect" in that light. The vast majority of the material nearly all of us listen to is 16/44.1. The vast majority of DACs made today will oversample that to 352.8, using three 2x steps. A few DACs will oversample to 8x rates in one step. A mere handful won't over sample at all. For this tiny handful that keep Redbook material bit perfect all the way through to the d/a conversion step, what is the result? Measurable, audible distortion. So we don't have fidelity. For fidelity, we have to oversample, in the DAC or prior. The oversampling offered by iZotope is better than most in-DAC oversampling, and can be done in one step rather than three. Simplicity - two fewer oversampling steps - in service of fidelity, i.e., better oversampling that results in better sound. Thus what looks at first impression like an add-on, oversampling in software, actually simplifies the process and results in better sound.

 

I have some ideas regarding ways that plugins may be able to help accomplish the same paradoxical feat: greater simplicity and fidelity brought by what looks at first glance like an add-on. But I'm using my laptop for music, and I'm tired of thumb-typing this on my phone, so I'll leave that for another time.

[Superdad]

I have some ideas regarding ways that plugins may be able to help accomplish the same paradoxical feat: greater simplicity and fidelity brought by what looks at first glance like an add-on. But I'm using my laptop for music, and I'm tired of thumb-typing this on my phone, so I'll leave that for another time.

 

I'll put forth one example you may be thinking of Jud: Loudspeaker crossovers. If someone really wanted to get into it, they could bi- or tri-amp their system, rip out the entire passive crossover, perform some measurements and design a crossover in the digital domain in software.

When the tools are there and I learn how to use them and have A+ use the plug-ins for this, I might try this myself. And mind you, I make my living selling film-and-foil capacitors (MusiCaps) to high-end audio electronics and loudspeaker manufacturers for their crossovers. There's no capacitor/inductor as good as having no capacitor/inductor (the custom woofers in my speakers have always run full range with no roll-off parts, and that helps them reproduce in a startlingly realistic way).

[mwheelerk]

While using A+ integrated with iTunes and playing from Smart Playlist in shuffle mode I have noticed a slight popping sound when the tracks change in both .5 and .6. Anyone else hearing a pop in between track changes?

[dmgbat]

Not to quibble, brlawyer, but this lawyer thinks that your position is almost wholly undermined by the statement you excerpted from Damien's white paper. In sum, Damien appears to argue that bit perfect is the beginning but not the end of digital audio reproduction. Based on what you excerpted above, I'd suggest that offering plug-ins, designed to combat that which is "detrimental [to] the sound quality," is entirely consistent w/ his stated goal.

 

Yes; we are on the same page, Roch.

 

I don't mind if the option is there for anyone to use it; but I just find it odd that, after years of preaching the opposite, we see a new "equalizer" in A+ (or other audiophile players for that matter). From Damien's own white paper:

 

"The computer is a great music server but also asource of jitter and other RF interferences thatare detrimental the sound quality, even whenbit-perfect reproduction is ensured.

 

The player software needs to optimize andstreamline the audio path to minimize theseadverse effects essentially linked to theprocessing load synchronous to the audiostreaming. Achieving “source direct” in additionto “bit-perfect” is key. "

 

 

 

 

You can obviously conclude from the above that the main if not only purpose of an audiophile-grade player like A+ has always been to ensure the most unadulterated reproduction of sound possible, i.e., "bit-perfect+source-direct".

 

Unless Damien can explain further, the use of AUnits defeats that principle completely, as it adds to the "User Space" the same distortions that "usual audio players" have when using Core Audio units.

 

Sorry for being too dogmatic here - it's just my lawyer's "consistent logic" mind working again.

 

p.s: On a totally unrelated note, does anyone know if my Nuforce Icon HDP is USB-asynchronous? I cannot find this information anywhere.

[Sam Lord]

There is a very simple solution: Don't use it.

 

However, DSP is a lot more accurate than an analogue equalizer, so the comparison is a bit silly. The point of bit-perfect playback is simply that nothing is done to the sound without your permission. The user should feel free.....

I agree with most of your points, except that in some very broad EQ applications some analog units still win. But that is mostly a production, not reproduction, matter. To make some recordings sound bigger and grander via the distortion of a hardware EQ, you should try out a Fearn if you ever get the chance. I think an excellent forum for these EQ questions is the PRW Mastering Forum and for room and speaker acoustics, the acoustics forum on PRW. The practical use and limits of room correction software is *in general* much better understood by those professionals, IMO.

[Encore]

Next time the problem recurs, select "Debug Info..." from the Audirvana Plus menu.

 

--David

 

Thanks. Will do ...

[philouu]

Hello

Some of you were mentioning ACR System 2, FuzzMeasure or FabFilter... to optimise sound in the digital domain (digital room correction) using the new AU feature.

Could you give more details about how you have integrated it with A+ ? Are you satisfied ?

thanks a lot in advance

philouu

 

Hello

Is there anybody having done digital room correction and using AU Plugins for the room correction within A+ ?

Thanks

philouu

[DigiPete]

A+ sounds and works fine with 5.1 PCM packaged in WAV and FLAC up to 24/96

 

Problems found:

 

 

1) FLAC 5.1 24/192 plays completely distorted and makes A+ unstable

2) The sorting mechnism does not work - klick # or Artist and nothing happens.

 

 

PS. Decibel is my usual player, I hear no difference in my rig.

[petecare]

Are those plugins available only in 32bit ? In this case you can try to launch A+ in 32bit mode to check. ... and inquire with TC Electronic about a 64bit release.

 

Bonne journée,

Damien

Thanks you for your quicks replies (Mytek beta firmware & AU TC Electronic).

 

At this moment I had forgotten mytek_stereo192-dsd-dac_firmware_V1.7.3.b2.

I'm staying with 1.7.1 version.

 

I have try A+ in 32 bits. TC Electronic AU are listed in this mode.

For sure, your soluce had been, so, mentioned by Apple (Using Logic Pro). And seems to solve this issue.

What difference with A+ mode 32 and 64 ??

 

But before my question, I tested Logic Pro running in 64 bits with the TC AU checked and passed test...

 

Thanks again to your efforts to maintain your soft with improvement and sound quality

petecare

[Nico]

With 1.4.9.7 I can not play FLAC 24/96000 single channel files. Heavy stuttering occurs besides the music in the background.

I have no problem at all to play the same files with 1.4.6.

[mwheelerk]

How do I stay on top of the latest versions of the beta. Whenever I check for updates I am told the version is current but then I notice a post in the thread that mentions a new version (such as above with 1.4.9.7) and I scramble to figure out where to get it. I go to Audivrana's site, their blog etc but struggle to find it. I did get this latest version by going back to the link in the first post of this thread.

[sidssp]

How do I stay on top of the latest versions of the beta. Whenever I check for updates I am told the version is current but then I notice a post in the thread that mentions a new version (such as above with 1.4.9.7) and I scramble to figure out where to get it. I go to Audivrana's site, their blog etc but struggle to find it. I did get this latest version by going back to the link in the first post of this thread.

 

You can't use check update for beta builds. You need to use the link Damien provided in this thread.

[wgscott]

Sorry for being too dogmatic here - it's just my lawyer's "consistent logic" mind working again.

 

I would agree with the "dogmatic" part of the self-assessment; "consistent" and "logic" -- perhaps less so.

[Cycleman]

I've just released 1.4.9.7 that brings:

• Audio Units presets load/save
  
• Option to limit bitdepth to 24bit for bridges the offer both 24 and 32bit bitdepth
  
• Fix issue leading to crash in Direct Mode on some Bluetooth devices
  
• and other minor fixes
  

 

Remains the per sample rate upsampling settings to have the feature set for 1.5

 

It is available at the same place: Download Audirvana Plus BETA | Audirvana

 

Hi Damien,

 

What's your sense of timing, for when the final 1.5 version will be released?

 

Also, is the current user's guide for 1.5 that comes in the download not fully done yet either?

 

I've been delighted with 1.4.9.7, with richer sound compared to the 1.4.6 version I had been (sometimes) using.

 

Dave, who also says 1.4.9.7 has worked no prob at all with the file types and equipment listed in his sig below...totally stable