HQ Player

[Hammer]

6 hours ago, Miska said:

 

If the input is from a file and output to a NAA, it is less problematic because NAA is handling the actual audio I/O with a large FIFO buffer.

 

From sound quality point of view it doesn't matter, unless you are getting drop-outs or other glitches. But it just makes things quite a bit less efficient, running an OS on top of another OS.

 

But why use virtual machine instead of running HQPlayer on the host OS?

 

Thanks...it’s because I am running everything on top of ESXi.  

[copy_of_a]

On 12/19/2018 at 6:01 PM, Miska said:

Pro

HQPlayer to convert files with your marvelous filters?!!

I'm sold!

... almost...

 

1.) with the Beta no meta data at all is transferred to the processed files (source FLAC or AIFF - target AIFF).

I think passing through embedded meta data and cover art is a must.

 

2.) on my machine (Mac Pro, OSX 10.11.6) the interface does not show a "record" button as shown in the manual (don't really need that function for the time being ... just noticed it's missing).

 

3.) for the Pro version I would suggest to place the min/max volume right in the main interface, not in the preferences. Too, I think it should be possible to bypass the built in limiter

 

4.) for conversions it would be extremely useful if we could set the source folder automatically as destination folder (so the drop down menu for "destination URI" should provide a preset for this). Even better if HQPpro could also create a subfolder as target (name it HQP processed or whatever) within the source folder for the conversions (handy if the output file extension is the same as the input file extension and you don't want to rename the processed files afterwards). Such an option is present in many DAWs and since HQP Pro is targeted at towards Audio Production Houses and such this might be a feature quite some users will aks for.

 

Really looking very foward to using HQPpro - great news ... and many thanks for making this great piece of software available for these purposes!!

 

[Miska]

12 minutes ago, Hammer said:

Thanks...it’s because I am running everything on top of ESXi.  

 

OK, so it is essentially Linux with customizations. But I don't pretend to understand why...

 

I prefer to run HQPlayer Embedded a bit like ESXi itself, on minimal OS, as close to the metal as possible.

 

[Miska]

18 minutes ago, copy_of_a said:

1.) with the Beta no meta data at all is transferred to the processed files (source FLAC or AIFF - target AIFF).

I think passing through embedded meta data and cover art is a must.

 

That is one of the hard cases. If you do AIFF to AIFF, or DSF to AIFF, or AIFF to DSF, (or anything else than FLAC) you get 1:1 copy of the metadata. In next release I plan to have FLAC to FLAC copy. But since FLAC uses different metadata format (Vorbis comments) than the others (ID3v2), there needs to be translation between the different metadata formats when you go to/from other format vs FLAC. That will be never be perfect copy and very tricky to get right except for the most basic items.

 

18 minutes ago, copy_of_a said:

2.) on my machine (Mac Pro, OSX 10.11.6) the interface does not show a "record" button as shown in the manual (don't really need that function for the time being ... just noticed it's missing).

 

There's no record button if you look closely, the look and feel depends on the OS. And on Linux (where the screenshots are from) it depends on the desktop environment and distribution's own theming. You can check the manual, recording happens in a different way. Plain record-button is not enough because you need to specify both input and output format (PCM/SDM, sample rate, etc). This is not as simple as in your typical recording application...

 

What you see in the screenshots is from Ubuntu Studio 18.04's theme engine.

 

18 minutes ago, copy_of_a said:

3.) for the Pro version I would suggest to place the min/max volume right in the main interface, not in the preferences. Too, I think it should be possible to bypass the built in limiter

 

I'm not too eager to clutter the main window with too many things. On other applications you usually cannot even change predefined ranges, even less so for hardware devices like mixing consoles.

 

If I allow bypassing the limiter, what do you think should happen when you exceed 0 dBFS?

 

18 minutes ago, copy_of_a said:

4.) for conversions it would be extremely useful if we could set the source folder automatically as destination folder (so the drop down menu for "destination URI" should provide a preset for this). Even better if HQPpro could also create a subfolder as target (name it HQP processed or whatever) within the source folder for the conversions (handy if the output file extension is the same as the input file extension and you don't want to rename the processed files afterwards). Such an option is present in many DAWs and since HQP Pro is targeted at towards Audio Production Houses and such this might be a feature quite some users will aks for.

 

I would like to avoid case where someone overwrites the source files with processed ones... I would also like to avoid need to rename the files afterwards because it is usually PITA. But I could make it create destination sub-folder automatically when source and destination are the same.

 

Since the main use is not to convert between for example AIFF and FLAC, but instead convert for example 5.1 channel DSD256 files to 5.1 channel DSD64, stereo 192/24 PCM, 96/24 PCM and 44.1/16 PCM the destination naming is slightly different case from the applications that are intended to convert from WAV to FLAC for example.

 

[freesteve]

On my Mac I record with Audio Hijack... sounds great ...

[trcns]

Just curious:

Is it possible to feed 44.1 bit rate CD ripped music files to HQPro and create 192 bit file or higher or some wide spactrum flac files which sound much better than upsampling in HQ Desktop version?

[Miska]

18 minutes ago, trcns said:

Is it possible to feed 44.1 bit rate CD ripped music files to HQPro and create 192 bit file or higher or some wide spactrum flac files which sound much better than upsampling in HQ Desktop version?

 

Yes, but algorithms are the same, so you get the same sound. You can also do same playback with Pro as with Desktop, without saving the output to a file too.

 

[copy_of_a]

1 hour ago, Miska said:

That is one of the hard cases. If you do AIFF to AIFF, or DSF to AIFF, or AIFF to DSF, (or anything else than FLAC) you get 1:1 copy of the metadata.

Miska... I was talking about AIFF to AIFF - no Meta data transferred at all here!

 

1 hour ago, Miska said:

What you see in the screenshots is from Ubuntu Studio 18.04's theme engine.

ok, I see ?

 

1 hour ago, Miska said:

I'm not too eager to clutter the main window with too many things. On other applications you usually cannot even change predefined ranges, even less so for hardware devices like mixing consoles.

If I allow bypassing the limiter, what do you think should happen when you exceed 0 dBFS?

Understandable! Than again my view is that Pros - for sure! - will be free and able to use their own (sometimes extremely expensive) limiters. If you are targeting your "Pro" version at the Pro community you have to provide flexibility as every engineer has his own workflow and flavours. Theres nothing wrong with a file that contains +4db clipping (or much more) intermediately as long as you don't do D/A. You can just as well tame down the peaks and intersamples later in the workflow with other tools if need is. Depends on the workflow...

In addition... some might want to upsample certain sound files or entire instrument tracks just to let their plugins run at higher rates. Especially "warmth" and "saturation" plugins that alias as hell sometimes but provide only medicore oversampling options (admittedly that's a special case).

Finally ... most quality DACs can handle moderate intersample peaks quite well... if I'm not mistaken.

 

1 hour ago, Miska said:

I would like to avoid case where someone overwrites the source files with processed ones... I would also like to avoid need to rename the files afterwards because it is usually PITA. But I could make it create destination sub-folder automatically when source and destination are the same.

My Idea was that I don't have to assign a destination folder at all. I simply set the source folder and choose the preset "create subfolder in source folder" by default and that's all. When playing around with your Beta it was quickly apparent to me that setting a destination folder for every source folder is just a waste of time. And e.g. from Steinberg Wavelab I am used to have such options which make live so much easier ...

[Miska]

32 minutes ago, copy_of_a said:

Miska... I was talking about AIFF to AIFF - no Meta data transferred at all here!

 

OK, strange, in my tests the ID3v2 data goes through just fine, I just tested again! Does it display the metadata like cover art and such in the main window?

 

32 minutes ago, copy_of_a said:

Theres nothing wrong with a file that contains +4db clipping (or much more) intermediately as long as you don't do D/A.

 

You cannot put that into output file, because it exceeds numerical range of the output samples! For example 16-bit PCM has range of -32768 - +32767 and there's no way you can encode bigger numbers in that 16-bit value.

 

And it is plain stupid to drive the peaks close to 0 dBFS, it is completely needless. One reason why we have stupid loudness wars. If you want to be safe with all DACs, you don't exceed -3 dBFS peak. You have plenty of headroom below if you output 24-bit, you could keep peaks at -20 dBFS and still you are nowhere near hitting the noise floor.

 

32 minutes ago, copy_of_a said:

You can just as well tame down the peaks and intersamples later in the workflow with other tools if need is.

 

My tool is supposed to be the last one that touches audio data before customer gets it...

 

32 minutes ago, copy_of_a said:

Finally ... most quality DACs can handle moderate intersample peaks quite well... if I'm not mistaken.

 

Not really. You better expect most DACs won't do that.

 

What I specifically don't want to allow is someone producing ugly digitally clipped crap with my software.

 

32 minutes ago, copy_of_a said:

My Idea was that I don't have to assign a destination folder at all. I simply set the source folder and choose the preset "create subfolder in source folder" by default and that's all. When playing around with your Beta it was quickly apparent to me that setting a destination folder for every source folder is just a waste of time. And e.g. from Steinberg Wavelab I am used to have such options which make live so much easier ...

 

I personally wouldn't want to mix final delivery files with the source files. Since in most cases final delivery files would be ZIPped it is most convenient if they are neatly in their own folders somewhere else.

 

Closest match for comparing software is probably Weiss Saracon.

 

P.S. Most DACs have lowest THD+N at -10 dBFS level.

[copy_of_a]

32 minutes ago, Miska said:

OK, strange, in my tests the ID3v2 data goes through just fine, I just tested again! Does it display the metadata like cover art and such in the main window?

yes, cover art is displayed (source)

 

32 minutes ago, Miska said:

ou cannot put that into output file, because it exceeds numerical range of the output samples! For example 16-bit PCM has range of -32768 - +32767 and there's no way you can encode bigger numbers in that 16-bit value.

ok... I was referring to 24 bit (actually to 32bit floating)

 

32 minutes ago, Miska said:

And it is plain stupid to drive the peaks close to 0 dBFS, it is completely needless. One reason why we have stupid loudness wars. If you want to be safe with all DACs, you don't exceed -3 dBFS peak. You have plenty of headroom below if you output 24-bit, you could keep peaks at -20 dBFS and still you are nowhere near hitting the noise floor.

agreed! but that's quite academical ... reality is different. And you won't change it with your offer.

Best recordings/masters I've personally purchased lately are Steven Wilson - To The Bone (2017) and The Pineapple Thief - Dissolution (2018). Those may respect your claims to some degree. But that is pretty much niche music for a really small target group (not only sound-wise). Personally I do adjust everything to -5db for listening ...

Todays Masters in popular music all touch 0dbfs (digital). I sure don't have to tell where those peaks go with minimum or intermediate phase upsampling...

 

32 minutes ago, Miska said:

My tool is supposed to be the last one that touches audio data before customer gets it...

ok! But that makes it more or less a "consumer" tool. Definitley a mature consumer tool at super high quality, but still.

With "Pro" I do associate a certain knowledge to take care of possible drawbacks with certain workflows (e.g. clipping). If you design your tool to take care of such things it's more or less targeted at the users of this community here at CA -- not so much at the Pro Audio world (mixing/mastering/file delivery). Which is really just fine since it's your product!

 

Anyway... I will defintely use it and am looking forward to the actual release!

 

 

[Miska]

21 minutes ago, copy_of_a said:

yes, cover art is displayed (source)

 

Is it embedded or detached cover art? How about album/song names and such?

 

21 minutes ago, copy_of_a said:

ok... I was referring to 24 bit (actually to 32bit floating)

 

Well, 24-bit or 32-bit integer have same limits. No way to exceed 0 dBFS, numbers are just bigger, but 0 dBFS is always the biggest number they can represent, dynamic range just extends downwards.

 

HQPlayer Pro doesn't allow file output in floating point, which does allow wider range because it is usually normalized in the middle. You cannot store floating point in FLAC, DSF or DSDIFF anyway. And I have never heard anybody selling downloads in floating point formats.

 

HQPlayer (all versions) happily read 32-bit and 64-bit floating point formats though (WAV).

 

By the way, I personally consider 32-bit float insufficient resolution for intermediate work files. They really should be 64-bit float.

 

21 minutes ago, copy_of_a said:

Those may respect your claims to some degree. But that is pretty much niche music for a really small target group (not only sound-wise).

 

Probably I'm anyway targeting niche group. I doubt many others would be using competing products (like Saracon) either. They just don't care enough. Most likely target group are those who deal with DSD or use DXD in Pyramix.

 

21 minutes ago, copy_of_a said:

But that makes it more or less a "consumer" tool.

 

Why? I don't think so. If you need to produce 44.1/16 FLAC for downloads from 192/24 source. Or if you produced your content in DSD256 but also want to release it on CD?

 

21 minutes ago, copy_of_a said:

With "Pro" I do associate a certain knowledge to take care of possible drawbacks with certain workflows (e.g. clipping).

 

Looking at how much content pro people put out that contains digital clipping, they obviously don't know how to take care of fidelity.

 

All HQPlayer products try their best to retain the best possible fidelity always. Clipping is the most common one ruining fidelity, so it needs to be controlled and prevented.

 

[Miska]

I'm not actually sure how many others support recording from DSD256 ADC on all three platforms, Windows, macOS and Linux? Many of the other tools I've got don't even support 768/32 ADC on these platforms, which is another format I use.

 

[copy_of_a]

34 minutes ago, Miska said:

 

Is it embedded or detached cover art? How about album/song names and such?

Embedded Album Art. Artist and Song Name get's displayed correctly while playing the source file... but the meta dta simply doesn't get transferred to the processed target file.

 

34 minutes ago, Miska said:

Well, 24-bit or 32-bit integer have same limits. No way to exceed 0 dBFS, numbers are just bigger, but 0 dBFS is always the biggest number they can represent, dynamic range just extends downwards.

since I am not really familar with these things I can't comment on the technical side or on the specs. All I know (hopefully) is that inside a DAW (most working in 32bit floating) you can push peaks and levels really far (+24db if I'm not mistaken) intermediately as long as you take care of leveling/limiting before "printing" (boucning/rendering) the file. To my knowlege vst plugins also provide this headroom. Not really sure about the specs here, though.

 

Quote

By the way, I personally consider 32-bit float insufficient resolution for intermediate work files. They really should be 64-bit float.

Plugins typically work at 64bit... some at 80bit. Only the DAWs work at 32bit float internally (which is intermediate anyway).

 

Quote

Probably I'm anyway targeting niche group. I doubt many others would be using competing products (like Saracon) either. They just don't care enough. Most likely target group are those who deal with DSD or use DXD in Pyramix.

ok, got it!

 

Quote

Why? I don't think so. If you need to produce 44.1/16 FLAC for downloads from 192/24 source.

In this case you probably use HQPpro as a resampler (only) and still check the files in the DAW... and maybe make a final touch-up there. In any case an engeneer would like to use his own limiter (even if it just gets the intersample peaks straight)... not a limiter in the conversion/resampler tool that can't be bypassed.

 

Quote

Looking at how much content pro people put out that contains digital clipping, they obviously don't know how to take care of fidelity.

?... well, yes, I agree ...

 

Quote

All HQPlayer products try their best to retain the best possible fidelity always. Clipping is the most common one ruining fidelity, so it needs to be controlled and prevented.

Absolutely! But all I am saying is that this should be an automated fix under the hood for playback. For conversion I whished it was an option (to bypass the limiter ... because I want to use my own beloved well known trusty workhorse limiter... if any at all).

 

I understand where you're coming from. I was maybe just reading too much in the term "pro" (which, for me, is the opposite of auto-anything under the hood).

 

 

[alec_eiffel]

I have asked the following question on the Exasound thread but got no answer, so I am reposting here:

I have tested my “new” E28 in stereo mode on my second system and I admit it sounds really really good, very musical, clearly ahead of another (and cheaper) ES9018 DAC I own, from Audiophonics.

 

The end game is to use all channels of the E28 in a 2 x 4 way setup. Room correction, active filtering, upsampling and volume control will be done in software by Roon or HQPlayer, over a stereo PCM stream (I don’t own a lot of DSD).

 

As all will be done in software I’d like to avoid as much as possible processing in the E28. WHat is your recommandation ? I read setting the volume to 0dB deactivates volume control, what about the best format/sampling rate to “match” the hyperthread DAC of the ES9028 ? 384kHz ? DSD64 ? DSD128 ? (DSD256 over 8 channel would probably be out of reach for my processing PC)

 

Thx!

 

Doe anyone knows what's the "best format" to use in HQPlayer to override the Exasound E28 filters?

 

Thx

 

[Bob Stern]

Hoping to avoid Linux, I’m trying to pipe the output of the Idagio app to HQP Desktop using an icecast server.  A Mac app called Ladiocast performs the icecast source encoding in either Ogg PCM or Ogg Vorbis format.

 

I used the following hqp-control command:

/Applications/HQPlayerDesktop.app/Contents/MacOS/hqp-control 10.0.0.5 http://10.0.0.6:8000/bob_stream

 

HQP Desktop does not play the music, but produces the following messages in the log:

 

Encoding:

Ogg PCM, bit rate 1411 kb/s (constant, average or variable)

 

HQ Desktop log:

& 2018/12/20 16:18:52 Playlist add URI: http://10.0.4.6:8000/bob_stream

# 2018/12/20 16:18:53 clPlaylist::AddURI(): unknown mime type: audio/ogg; codecs=pcm

 

Encoding:

Ogg Vorbis (variable bit rate 1411 kb/s)

 

HQ Desktop log:

& 2018/12/20 16:25:01 Playlist clear

& 2018/12/20 16:25:01 Stop...

  2018/12/20 16:25:01 Close previous transport

& 2018/12/20 16:25:01 Set transport (240):

  2018/12/20 16:25:01 Automatic output rate switching enabled

& 2018/12/20 16:25:01 Playlist add URI: http://10.0.4.6:8000/bob_stream

# 2018/12/20 16:25:02 ReadFLACErrorCB(): lost sync

# 2018/12/20 16:25:02 ReadFLACErrorCB(): lost sync

# 2018/12/20 16:25:02 ReadFLACErrorCB(): lost sync

# 2018/12/20 16:25:02 ReadFLACErrorCB(): lost sync

# 2018/12/20 16:25:02 ReadFLACErrorCB(): lost sync

# 2018/12/20 16:25:02 ReadFLACErrorCB(): lost sync

# 2018/12/20 16:25:02 clPlaylist::AddURI("http://10.0.4.6:8000/bob_stream"): clAudioTransport::GetTrackLength(): uSampleRate == 0

 

(Actually, the lost sync entries are repeated many more times before the clPlaylist entry.)

[Miska]

36 minutes ago, copy_of_a said:

Embedded Album Art. Artist and Song Name get's displayed correctly while playing the source file... but the meta dta simply doesn't get transferred to the processed target file.

 

OK, no idea then... It is working for me, I'd need to be able to reproduce this somehow.

 

36 minutes ago, copy_of_a said:

since I am not really familar with these things I can't comment on the technical side or on the specs. All I know (hopefully) is that inside a DAW (most working in 32bit floating) you can push peaks and levels really far (+24db if I'm not mistaken) intermediately as long as you take care of leveling/limiting before "printing" (bouning/rendering) the file. To my knowlege vst plugins also provide this headroom. Not really sure about the specs here, though.

 

Yes, same with HQPlayer Pro, you could have peaks even 200+ dB internally, no problem. But for "rendering" the output file it needs to fit in the output format. That's where the limiter kicks in. HQPlayer works in 64-bit floating point. Some selected places use 80-bit floating point.

 

So no real difference here, the only difference may be what happens when you don't obey the rule but try to render out +24 dB or something. If you don't take care of the leveling, HQPlayer does it for you in it's own way. As long as you take care of it properly, nothing happens.

 

36 minutes ago, copy_of_a said:

Plugins typically work at 64bit... some at 80bit. Only the DAWs work at 32bit float internally (which is intermediate anyway).

 

Yeah, but I mean work files you store on a disk, the non-final ones.

 

36 minutes ago, copy_of_a said:

In this case you probably use HQPpro as a resampler (only) and still check the files in the DAW... and maybe make a final touch-up there. In any case an engeneer would like to use his own limiter (even if it just gets the intersample peaks straight)... not a limiter in the conversion/resampler tool that can't be bypassed.

 

You do final touch-up in DAW, and then as last thing you convert the final delivery files with HQPlayer. Not the other way around, because otherwise you again destroy fidelity.

 

36 minutes ago, copy_of_a said:

Absolutely! But all I am saying is that this should be an automated fix under the hood for playback. For conversion I whished it was an option (to bypass the limiter ... because I want to use my own beloved well known trusty workhorse limiter... if any at all).

 

As I said earlier, you need to limit the output somehow when you are trying to exceed possible value range of the samples. There's no way around it, you cannot put 2 liters of cola in 1 liter bottle, if you try, it will spill over one way or the other. And it doesn't remain secret, HQPlayer keeps incrementing the counter and turning the volume knob red whenever the limiting is triggered.

 

Volume knob also turns yellow at -1 dB and red at 0 dB level settings to indicate that you are in imminent danger of getting triggers (Limited-counter will the actual situation).

[Miska]

13 minutes ago, alec_eiffel said:

I have asked the following question on the Exasound thread but got no answer, so I am reposting here:

 

Doe anyone knows what's the "best format" to use in HQPlayer to override the Exasound E28 filters?

 

DSD256, either ASDM5 or ASDM7. Choice of filter is more depending on music and personal preferences. It is nice DAC also in a way that it (IIRC) supports also 48k-base DSD rates.

 

Doing DSD256 for e28's all 8 channels is quite heavy, that's why I got the monster machine for it.  Stereo is not a problem though.

 

If you don't quite have the oomph for DSD256 at 8 channels, you can do PCM at 384k and 32-bit TPDF dithered output as alternative.

 

[Bob Stern]

11 minutes ago, alec_eiffel said:

what's the "best format" to use in HQPlayer to override the Exasound E28 filters?

 

Here are my settings based on Jussi's general recommendations, not specific to exaSound DACs:

[Miska]

6 minutes ago, Bob Stern said:

Hoping to avoid Linux, I’m trying to pipe the output of the Idagio app to HQP Desktop using an icecast server.  A Mac app called Ladiocast performs the icecast source encoding in either Ogg PCM or Ogg Vorbis format.

 

I used the following hqp-control command:

/Applications/HQPlayerDesktop.app/Contents/MacOS/hqp-control 10.0.0.5 http://10.0.0.6:8000/bob_stream

 

HQP Desktop does not play the music, but produces the following messages in the log:

 

Encoding:

Ogg PCM, bit rate 1411 kb/s (constant, average or variable)

 

HQ Desktop log:

& 2018/12/20 16:18:52 Playlist add URI: http://10.0.4.6:8000/bob_stream

# 2018/12/20 16:18:53 clPlaylist::AddURI(): unknown mime type: audio/ogg; codecs=pcm

 

The problem here is that it is trying to send Vorbis encoded (lossy codec similar to MP3) data and Vorbis (or MP3) is not supported, at least not at the moment.

 

That Ogg PCM thing maybe fixable, but it is first time I'm seeing such, need to check out.

 

Plain FLAC or Ogg-FLAC both should work. Same for WAV/AIFF too.

 

[Bob Stern]

Ogg PCM is the only option offered by Ladiocast.  No FLAC, AIFF or WAV.

[Miska]

5 minutes ago, Bob Stern said:

Ogg PCM is the only option offered by Ladiocast.  No FLAC, AIFF or WAV.

 

The notice on top of the spec is not very promising:

https://wiki.xiph.org/OggPCM

 

[Geoffrey Armstrong]

12 hours ago, Miska said:

 

I'll send you email...

 

 

You mean Desktop couldn't access the DAC? Probably there was something left running of Pro to keep the DAC busy. You could check task manager if the exe is still shown running and kill it if it is hanging there for some reason. They share the same configuration, but the configuration is cross-compatible. I do the same thing quite a lot myself all the time.

 

If you just want to convert file, use the Null Output mode instead (PCM N/O or SDM N/O). This will leave the DAC alone when operating, and then you are not bound anymore by the formats supported by DAC.

 

Thanks.

 

Yes, desktop couldn't access the DAC until after a reboot.

 

OK, will check Task Manager next time.

 

I did use the null output mode; but used the DAC for the input mode. I figured this is what I needed to do to have access to the filters, modulators and sample rates supported by the T+A dac to convert the input source file using poly-sinc-xtr with ASDM7 to 48X512.

 

Geoff

[bibo01]

13 hours ago, Miska said:

 

I'll send you email...

 

 

Can you send that email to me too?! 

[Miska]

39 minutes ago, Geoffrey Armstrong said:

I did use the null output mode; but used the DAC for the input mode. I figured this is what I needed to do to have access to the filters, modulators and sample rates supported by the T+A dac to convert the input source file using poly-sinc-xtr with ASDM7 to 48X512.

 

Input doesn't matter in that picture, those are defined by the output. Input device is only used when you enter the input URI as source, such as "audio:default/705600/2". HQPlayer itself can convert anything to anything, but DACs naturally impose limits on what they can accept. Null Output lifts those limitations by exposing output that supports everything.

 

[tranfa]

I'm testing the Pro too.

 

I'm having an odd behaviour coming to the meters: after the first closing of the system, I can't visualize them anymore...and I don't think I changed much in the settings.

 

Also, coming to the conversion, I'm not sure I fully understood the limitations of the evaluation version.