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[teroyk]

I actually looking for older Waveburner that doesn't use floating point.

What does it use then?

At least some old web page did not said that it use 32-bit floating point.

[teroyk]

especially when you have to convert it to back to integer (for play with all audio interfaces).

Sigma-Delta DACs can be designed to operate directly on floating point numbers. I think the Burr-Brown DAC in the TiBook can do it.

That sounds interesting...I want know more about that.

[teroyk]

but adding stereo/split stereo/mono and eventually filetypes could be interesting.

Yes that will be usefull too, because there is sometimes interesting incompability problems. And filetypes, because it is not long time when thinking what program can open Broadcasting Wave (WAV with time code, sometimes .BWV or .BWAV).

[IIO]

And what is the correct way of doing it?

you have two options, if you use 24 bit aritmetics you first multiply with a factor and then divide again by something like 256, the other option is to use float, but only allow values to be set which equal/end up in exact 24  bit values.

the first version requires overflow bits, like present in the 24.8 stream format of old protools sytems and the very first iOS.

since a GUI slider also has a limited resolution you should not loose anything interesting by doing that.
 
 
the idea to use plug-ins when using waveburner as music player is a misconception imho.

if you play a CD track from a commercial release you are playing a 16 bit master which might already be dithered down from whatever. such masters are supposed to be sent directly into the DAC and not processed again.

since he uses his computer as fileplayer, he is well advised to not change the volume at all. that is what amplifiers are for. just think "CD player". CD players do not have in internal, digital volume regulator either.

if he wants to become an audiophile listening expert these theories are more important than expensive cables.

[IIO]

Yes that will be usefull too, because there is sometimes interesting incompability problems. And filetypes

for my part i use aif only. SDII was already legacy when i started recroding in 1992 and waves is a windows thing.

the various issues and options of the wave, wave64 and whatnotwave formats on windows are still a mystery to me.

you are still right that it should be included to make the thing complete.

[nateman831]

WaveBurner Pro will do all of the above

WaveBurner Pro sounds interesting to me... Looks like the app is pretty hard to track down though. I've tried searching a number of sites, but come up empty-handed. As best as I could figure out, v2.2.1 would have been the latest version for OS 9, but I'm not even sure of that.

If you're seriously concerned about high quality playback, i.e. the real audiophile playback, you should consider doing it in OSX and, preferably, on Intel Mac.

Feel free to share any details on how you feel it shines. Or what set up is needed to make it shine. I'm always open to learning, although I also know opinions vary from person to person on what constitutes "audiophile" sound.

I've actually tried Mac OS X many times over the years. Mostly on Intel processors. Unfortunately, to me, it was a let down. I sense we may have different primary factors when it comes to what constitutes audiophile sound.

For me, having an unbloated OS is a primary consideration, and this project (ASIO on Mac OS 9) makes sense in that context.

My initial impressions of OS X actually were positive. I felt it had a more natural audio presentation than either Windows XP or Windows 7 did.

And I can understand why that presentation might be a primary factor for some people. Unfortunately, with more listening, it was a let down for me because of the greater "bloat" of OS X compared to other OS'es like Windows. This is something I've pinpointed based on lots of testing and listening to music across every platform: Windows, Mac OS X, Linux, iOS, Mac OS 9, etc - the rule holds true for me everywhere. If you open Activity Monitor on OS X and Task Manager on Windows, you can compare the number of running threads and processes and Mac OS X has more "congestion". But I was already hearing that in the music before I looked it up. The 2nd issue is a little more ambigious. On OS X the audiophile players can utilize something called "exclusive mode" and that's the only high end option. Why is that the only option? How does it compare (on a technical level) to options in Windows such as ASIO, WASAPI, or Kernel Streaming? Without really knowing the details, it comes across to me as a lack of serious development on the OS X side.

Anyways, I found I preferred any of those 3 options in Windows to exclusive mode in OS X, but it's hard to tell if that's just because of the bloat in OS X. Then I found I usually prefer ASIO over all the others, and Mac OS 9 happens to have that.

As far as the "bloat" issue goes; it's a primary factor to my musical enjoyment. Through lots of listening, I've learned that I enjoy music a lot more on lighter OS'es. When the OS is really lean (less going on in the background; more dedicated only to music), I tend to "just enjoy the music" whereas when the OS is bloated I tend to get bored and get listening fatigue really quickly. Then I usually change tracks a lot and end up listening to very narrow genres. I find I can be more forgiving of other shortcomings when I have this one thing, as long as the other things aren't too bad.

I also notice other benefits of a lighter OS as well: more fine details, more atmosphere, better "blackground", etc. many of these differences show up especially in longer listening sessions after my ears settle in. But I always pay attention to the boredom vs engagement factor and the biggest determination of that for me is having a leaner OS.

Anyways, having said all that - now you know where I'm coming from.

I'd definitely be open in hearing what you appreciate about OS X for music playback, and what steps you recommend to get the most out of it.

[ssp3]

WaveBurner Pro sounds interesting to me... Looks like the app is pretty hard to track down though. I've tried searching a number of sites, but come up empty-handed. As best as I could figure out, v2.2.1 would have been the latest version for OS 9, but I'm not even sure of that.

Not that hard. And yes, 2.2.1 was the last version for OS9
http://macos9lives.com/smforum/index.php/topic,6894.0.html

Feel free to share any details on how you feel it shines.

Oh, thank you!

Or what set up is needed to make it shine.

In short, start with proper listening enviroment and then use something close to what mastering engineers use in their studios.

I sense we may have different primary factors when it comes to what constitutes audiophile sound.

Yes, we do. Dynamic headphones is not audiophile sound in my book, for example. Hi-Fi at best. When I use the headphones (if I have to), I use Stax Lambda Signature with SRM-T1 tubed driver unit. Shall I continue?

For me, having an unbloated OS is a primary consideration, and this project (ASIO on Mac OS 9) makes sense in that context.

Bloated or not bloated is of secondary concern. More important is whether:
* the particular ASIO driver is bit transparent
* the playback application in question is correctly written and bit transparent (Peak/ASIO for AMIII combo, for example, is not)
* the master clock is jitter free, does not get polluted over the air or power supply lines by digital garbage generated by processors, other ICs or PSU and where it (clock, that is) resides.
Ideally it should be near the D/A converter chip and the transport (computer) slaved to it.

Unnecessary processes/kexts in OSX can be unloaded, if they bother you. Just like extensions in OS9.

Without really knowing the details, it comes across to me as a lack of serious development on the OS X side.

One has to wonder how mastering engineers have managed to churn out thousands and thousands of CDs and HD releases on such inferior platform using such inferior tools. Most of them must be deaf, no? 

[IIO]

although I also know opinions vary from person to person on what constitutes "audiophile" sound.

it is above all not the operating system or the application you use.

and hifi guys and musicians will always disagree about various stuff, because they often do not know the other parties world  or gear.

you want an OS9 based "audiophile" setup recommendation ?

 - professional room damping to the max

 - and ideal listening spot where you sit

 - full range monitor speakers >3000 USD

 - amplifier type does not matter imho, any good studio or hifi amp will do

 - good converter with proper analog parts, for OS9 you are probably left with mykerinos or hammerfall PCI based systems

 - the interface clocked by a big ben master clock

 - the big ben´s clock reclocked by a mutec reclocker before going into the DAC.

software side one might want to use restoration tools from waves for the one or other older recording, and case you listen to floating point a dithering with noise shaping of your choice. the player app itself is completely irrelevant, but should be able to host plug-ins for such cases as above.

[ssp3]

@IIO, what do you make of it? What kind of weirdness is it? Where all this stuff below 16th bit comes from?

Recent 2 track editing app. 16 bit playback. Master fader untouched at "0". No dither, no any kind of processing, just this bitscope plugin inserted as master effect.

[ssp3]

 - professional room - Yes

- damping to the max - No

 - and ideal listening spot where you sit - Agree

 - full range monitor speakers >3000 USD - Agree. What you pay is what you get

 - amplifier type does not matter imho, any good studio or hifi amp will do - Don't agree, it does matter big time

 - good converter with proper analog parts - Agree

 - the interface clocked by a big ben master clock - So so, it's still reconstructed by PLL. Oscillator is better.

 - the big ben´s clock reclocked by a mutec reclocker before going into the DAC. - 


As to the musicians, I'm always on their side - they shoud use whatever makes them groove. They just have to leave the final step to people in the trade who also have trained ears and proper listening enviroment.

[IIO]

the audiophile retards scene (we call them "cable hearer" in german) has put 4 of those reclockers in series, which allegedly makes a difference.

if you want to be on the safe side you should use 20 of course!
 
 
but jokes aside, if you own 7 vintage motu interfaces like me, and run up to 32 channels IO on one machine, it can be a great idea to get one of these mutec... with about 1000 euro it costs as much as 4 used interfaces and definetely makes the sound better.

[ssp3]

I had a friend who used to say something like this about the equipment he was selling:
"Du schliesst das Ding an, shaltest es ein und die Sonne geht auf" 


**For non-German speakers: "You connect this thing, switch it on and the sun begins to rise"

[IIO]

@IIO, what do you make of it? What kind of weirdness is it? Where all this stuff below 16th bit comes from?

not sure what you mean?

editor apps are for editing or creating audio, and back in the days i´ve often used 5 apps at the same time.

the more you want to edit files, the more it makes sense to remain in float, but half the apps do not support it. so you ended up converting or exporting things depending on what you plan to do next.

not exactly the definition of fun, but what else could you do about it.

[ssp3]

@IIO, what do you make of it? What kind of weirdness is it? Where all this stuff below 16th bit comes from?

not sure what you mean?

It's a 'bit transparency' test. Using 2 track editor as file player. 16 bit file is being played back. No dither, no any kind of processing, nothing. All prefs and possible hidden "features" checked and verified that nothing is switched on.

Monitoring active bits with bitscope plugin inserted into the master bus.

All similar apps under identical conditions show only 16 active bits, as it should be. This one shows activity below 16th bit. I'm not sure what could have caused it, hence the question(s).


Merry Chrismas, everyone!

[IIO]

ah, i did not understand that "below" because i did not look at the picture.

well, that can happen for a number of reasons. most of the time it wont matter when the actual converting happens, in other cases it might.

what is the exact setup here? coreaudio OSX should transport 16 bits as is and not invent additional patterns or fill with zeros.

but if an app has a VST interface, there we are in float already. that it looks strange does not mean that the information would not be preserved when converting back to int.

only 32 bit int might suffer from a float-and-back-conversion. (but i did not have coffee yet)

p.s.: a signal between -1 and 1 (which is what music in an integer resolution has) will occupy 50% of the possible range of 32 bit float. only over 0 db bit #22 and bit #30 are both true at the same time (but as i said i dit not had coffee yet)

[ssp3]

what is the exact setup here? coreaudio OSX should transport 16 bits as is and not invent additional patterns or fill with zeros.

It should, but this particular app doesn't. It's a Japanese cross-breed of Peak and Wavelab. All other similar apps behave normally under identical conditions.

but if an app has a VST interface, there we are in float already.

I'm not sure what do you mean by 'VST interface'. Most, if not all similar apps, have the means to load the VST plug-ins, but most behave as expected - garbage in, garbage out, no additives.

I did try another test setup without any VST plug-in on master bus, not even bitscope plug-in. The app streams out over FireWire to external MH interface. Then the patchbay there is set up to send the signal back to Spectrafoo analyzer.
The result is the same - all 32 bits are active when playing 16 bit file.

As someone once said - never turn your back to digital audio. It equally applies to OS9, Win, ASIO and other kind of audio drivers. Plenty of opportunities to screw something up.

[IIO]

All other similar apps behave normally under identical conditions.

we do not know what the condition in this app is.

for some weird japanese player app i would almost exspect that it can only forward float format to the VST interface, and then that viewer plug-in shows that stream. for all other plug-ins beside this one, thigns would be processed in float anyway, i.e. if the host would be capable of forwarding a 16 bit stream, the plug-in would convert it.

and it should not be a problem because the conversion from 16 to 32 and back is basically losless if you do not process the signal somehow.

the app just does not allow to monitor its 16 bit output with this plug-in.

make a test file of 5 samples and record it through that app. that is the only way to find out what really happens.

[ssp3]

the app just does not allow to monitor its 16 bit output with this plug-in.

IIO, you didn't read in full what I wrote. Again. 

I am not using the plug-in in my second test! I am monitoring the output stream of the app with another app. Still 32 active bits!
It is the same data stream that goes out to S/PDIF or AES/EBU digital outputs (via CoreAudio) and that I will be listening to on external DAC!
Will CoreAudio convert those 32 bits back to 16? If so, how? Chop off lower 16 bits? Is my audio still pristine and not manipulated? How can I be sure?
In audiophile world and in mastering data integrity is of paramount importance.

make a test file of 5 samples and record it through that app. that is the only way to find out what really happens.

Nah, I'm not going to waste my time. I simply don't trust this app. Nice looking, some useful features, like really good scrub or continuous backwards play at various speeds, but otherwise just weird junk. I have better tools at my disposal.

I was just giving an example on how easy it is to fall into trap by using nice looking+many features, but otherwise "broken"' app and then probably blame the OS for shitty sound. It can happen in OS9 too.

[IIO]

Still 32 active bits!
It is the same data stream that goes out to S/PDIF or AES/EBU digital outputs (via CoreAudio) and that I will be listening to on external DAC!

but that can´t be, at one point the float format must be converted to int in order to pass it to a s/d converter.

where this happens? no idea. that coreaudio itself can transport a format as is does not guarantee that apps send the stream as we wish.

from 10.5 on, OSX transports 32 and 64 bits in eithegr float or int, somewhen later it added PDM/DSD and whatnot.

Will CoreAudio convert those 32 bits back to 16? If so, how? Chop off lower 16 bits?

nope, the conversion between int and float has to happen arithmetically.

a floating point (or fixed point for that matter) stream has to be clipped to -1.0 to +0.999681 and then you can map the values it contains to the 16 bit signed range which is -32768 to 32767.

because float 32 allows millions of different values between -1 and 1 and 16 bit only has 64500, the conversion is literally losless.

float 32 allows always at least 5 decimal places of precision, so 16 bit values "fit" right into it.

the issue in practice is that our spark friend from above might perform a gain change in 32 bit when he uses a mac as virtual CD player. if you do that, the resulting float values will no longer match into the 16 bit scheme, and then you may use dithering to make things a bit better.

Is my audio still pristine and not manipulated? How can I be sure?

Nah, I'm not going to waste my time.

you will have to decide if you want to test it or not. no test, no evidence.

16 bit is long dead if you ask me. but of course we all have mpeg stuff somewhere in out collection and i also still have CDs....

[IIO]

enter anything between -30000 and +30000 in the first line.

or -85000000 to 85000000 for 24 bits.

https://www.h-schmidt.net/FloatConverter/IEEE754.html

what works for 30000 will also work for (30000/65535)