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Posted on Sun, Sep 15 2019 11:09
by llGoDLikEll
Joined on Fri, Aug 30 2019, Posts 3

Can't find the details in the pages or Manual

Posted on Sun, Sep 15 2019 19:40
by Dietz
Joined on Tue, Aug 06 2002, Vienna / Europe, Posts 7305

Welcome llGoDLikEll,

Vienna Instruments are recorded in 96 kHz with 24 bit resolution. All recordings are stored and processed in 32 bit FP resolution. In a final step, the samples are converted to 44.1 kHz / 24 bit just before mapping, using a proprietary SRC tool. To reduce their size, the data of the instruments is compressed (not reduced) before delivery.

HTH,

/Dietz - Vienna Symphonic Library
Posted on Mon, Sep 16 2019 04:31
by robinlb1900
Joined on Sat, Jun 22 2019, Posts 17
In the future, we expect to have a 48K/24bit version available for users to choose.
Some other VSTs provide it at present.
Of course, I may have to upgrade my SSD again.
Posted on Tue, Sep 17 2019 08:12
by llGoDLikEll
Joined on Fri, Aug 30 2019, Posts 3

Originally Posted by: Dietz Go to Quoted Post

Welcome llGoDLikEll,

Vienna Instruments are recorded in 96 kHz with 24 bit resolution. All recordings are stored and processed in 32 bit FP resolution. In a final step, the samples are converted to 44.1 kHz / 24 bit just before mapping, using a proprietary SRC tool. To reduce their size, the data of intruments is compressed (not reduced) before delivery.

HTH,

Thank you very much for the detailed description.

But it will be awesome if there is a 96k/24bit version for more precise work

Posted on Wed, Sep 18 2019 11:49
by robinlb1900
Joined on Sat, Jun 22 2019, Posts 17
Originally Posted by: llGoDLikEll Go to Quoted Post
Originally Posted by: Dietz Go to Quoted Post

Welcome<span style="color: #000000;">llGoDLikEll,
<span style="color: #000000;">Vienna Instruments are recorded in 96 kHz with 24 bit resolution. All recordingsare stored and processed in 32 bit FP resolution. In a final step, the samples are converted to 44.1 kHz / 24 bit just before mapping, using a proprietary SRC tool. To reduce their size, the data of intruments is compressed (not reduced) before delivery.
<span style="color: #000000;">HTH,

Thank you very much for thedetailed description.
But itwill be awesome if there is a 96k/24bit version for more precisework


I think so, but it will be a huge file:)
48k may be more realistic.
Of course, if there is 96k I will happy to prepare 1TB SSD</span></span></span>
Posted on Wed, Sep 18 2019 14:34
by PaoloT
Joined on Tue, Dec 27 2016, Posts 487

I have several of the libraries from Sonokinetic. They are so kind to supply not only the original 24 bit samples, but also the 16 bit version. I prefer to use this latter, because they are much lighter on the CPU and memory – and they sound the same to me!

Paolo

Posted on Wed, Sep 18 2019 21:00
by dmidi
Joined on Wed, Sep 18 2019, Posts 5

96 Khz @ 24 bit would be quite nice, and would bring lower latencies, which can be significant for playback 'feel'. Latencies are cumulative and live recording sampling rate is determined by the lowest rate of any part of the chain.  So the net effect of VSL sticking to a 1980's era sampling rate is a doubling of the latencies of every part of the playback chain.

A 2 TB NVME PCIe SSD is $200.  32 GB of DDR4 RAM for a laptop is $150. 

Not sure why VSL has decided to choke the sound chain latency by sticking to 44/48 Khz. 

Posted on Thu, Sep 19 2019 09:35
by PaoloT
Joined on Tue, Dec 27 2016, Posts 487
Actually, using 24/96 samples will slow down the system. Without adding any perceivable quality.

Paolo
Posted on Thu, Sep 19 2019 19:50
by dmidi
Joined on Wed, Sep 18 2019, Posts 5

Originally Posted by: PaoloT Go to Quoted Post
Actually, using 24/96 samples will slow down the system. Without adding any perceivable quality.
Paolo

There are 2 different ways to consider 'slowness'. 

1.  How slow is the computer hardware doing the processing, which is how you seem to consider this issue.  Of course, if one doubles the sample rate then processing power and bandwidth must also be double or the system will 'slow down'.  However, almost all laptops or desktops of a mid-grade specification (~$1000), and are less than 5 years old, already have excessive amounts of resources and will not be affected by 96 or even 192/384 Khz sampling rate.  Our hardware is fast enough to not be slowed by increasing sample frequency.

2.  What is affected by the other type of 'slowness' is the piano player having to deal with the lagtime or delay that occurs in the time period from pressing a key to hearing a sound; this is referred to as latency.  And latency absolutely affects the quality of the playing experience.  The lower the sampling rate the higher the latency, and the higher the latency/delay the more the player has to adjust to the different 'feel' and 'sound'. 

Certain players feel the affects of latency at 3 ms, and there have been many studies into psychoacoustics which support the idea that such low latency is perceived and affects musicians.  A 44 Khz sampling rate makes it more difficult to achieve good sound quality and low lagtime/latency in the instrument.

Given thier ownership of awesome acoustical grands, robotic sampling system for the same, access to a world class studio, and having ample technical resources, VSL is in an excellent position to release an engineering study on latency and include it in their product information and customer education materials.  It would be great to see data on the internal latency of the Synchron player, and then see how that sampling rate interacts with other parts of the processing chain and thereby affects total system latency.

If VSL doesn't do it...maybe I'll do it in the living room?  Or can I get an invite to Vienna?  ;)

On a concluding note, if one really wants to look at the necessity or value of the VSL library configuration, I would suggest that 24 bits of dynamic range for an acoustical instrument is ABSOLUTELY overkill when considering most scenarios.  24 bits equals 144 dB of dynamic range, and I dare say that even the most enthusiastic pianist can hardly play the Steinway 274 at a level that competes with jet engines at 100 feet or machine gun fire at 1 meter.  Of course, there is the convenience of laying down 24 bit tracks that coexist with live mics that do not have their gain properly adjusted, but that is convenience only, while 96+ Khz sample rate potentially has a much more fundamental impact on playability.  And of course, 24 bits does give MIXING headroom, but that is not required for SAMPLING of a grand piano, so VSL gave themselves much more headroom than could possibly be useful.  Even 16 bits is excessive dynamic range for a piano library.

BTW, Pianoteq Pro includes 96 Khz sampling rate and also includes the Steinway D in their library.  Could make for an interesting comparison between sampled and modeled sound.  FWIW, I think the VSL would come out ahead at the same sampling frequency, but would not prevail for live playability if having double the latency.

Given that VSL is recording at 96 Khz (with 32bits!) it really is amazing that there is no 96 Khz release that exists alongside the 44/48 Khz release.  Can we get an explanation for that decision and possibly some analysis of VSL's internal latency?

Posted on Thu, Sep 19 2019 20:05
by PaoloT
Joined on Tue, Dec 27 2016, Posts 487

But you are not forced to use 24/96 samples to set your audio interface to 24/96. You can still play lower resolution/frequency samples, and still enjoy the lower latency warranted by higher system settings.

I agree that nowadays computers are very powerful, but when dealing with all the instruments of an orchestra, maybe with multiple microphones, and additional processing, you'll end up with any system chocking. Going the for a an excellent 24/44.1 would give CD quality, while keeping the system relatively light.

Paolo

Posted on Thu, Sep 19 2019 20:28
by dmidi
Joined on Wed, Sep 18 2019, Posts 5

Originally Posted by: PaoloT Go to Quoted Post

But you are not forced to use 24/96 samples to set your audio interface to 24/96. You can still play lower resolution/frequency samples, and still enjoy the lower latency warranted by higher system settings.

Paolo

I agree that there are scenarios where higher sampling rate would impose excess processing or storage.  But the approach of increasing sampling rate after the VSL library actually increases latency, as upsampling the frequency rate requires another processing step.  

Better that VSL gives us the option of a higher sample rate library.  Then, even in scenarios requiring lower sample rates for recording the orchestra, etc, we could just use a few processor cycles to downsample the frequency rate without even increasing the latency (this only works for downsampling).

FWIW, I would like to see some double blind latency tests with varying freqency rate.  I would not be surprised at all if we have 'super hearers' that can process sound much faster than average people...and my guess is that some very gifted musicians might actually be able to process better with lower levels of latency. 

I am trying to systematically eliminate the negative effects on sound quailty and differences in playability between 'mechanically voiced' and 'electronically voiced' grand pianos.  VSL has potentially done much to eliminate potential negative effects, and the 96 Khz might just be another step towards bridging the divide.

Posted on Thu, Sep 19 2019 23:10
by Itchy
Joined on Sat, Feb 18 2017, Posts 30

Just my opinion, and I've been known to speak with my foot in my mouth many times, but this is my understanding (I earned my B.M. degree in Music Production and Engineering years ago when digital audio was new.)

You’re saying that systems these days are *optimized* for sample rates like 96KhZ @ 24-bits… to the point where input, processing, and output at that rate is actually *faster* than it would be if it had to digitally convert half as much data? Hmmm. You mention a few clock cycles to convert. I’m not a CPU expert nor an Apogee employee, but my understanding is that latency in one's system depends on I/O, processing, and conversion (at least). If I were a betting man I’d guess the nanoseconds it’d take interpolating half as much data would ballpark-equivalent to natively-processing twice as much data.

~Christopher
Posted on Fri, Sep 20 2019 01:46
by robinlb1900
Joined on Sat, Jun 22 2019, Posts 17
Agree with dmidi.
I suggest and look forward to VSL offering 96K/24bit option to users, at least for synchron piano products.
Then VSL can continue to stay ahead of sampling source.
Posted on Tue, Sep 24 2019 21:53
by littlewierdo
Joined on Sun, Apr 24 2016, Posts 221

Originally Posted by: Itchy Go to Quoted Post

Just my opinion, and I've been known to speak with my foot in my mouth many times, but this is my understanding (I earned my B.M. degree in Music Production and Engineering years ago when digital audio was new.)

You’re saying that systems these days are *optimized* for sample rates like 96KhZ @ 24-bits… to the point where input, processing, and output at that rate is actually *faster* than it would be if it had to digitally convert half as much data? Hmmm. You mention a few clock cycles to convert. I’m not a CPU expert nor an Apogee employee, but my understanding is that latency in one's system depends on I/O, processing, and conversion (at least). If I were a betting man I’d guess the nanoseconds it’d take interpolating half as much data would ballpark-equivalent to natively-processing twice as much data.

Not to mention the memory footprint. A full orchestral piece requires 60 plus gigabytes of ram for me, and that is really nothing in comparison with what it could be. There is a reason why 44.1 at a 16 bit depth rate was chosen for CD audio, this is considered just above the maximum quality that anyone can hear a difference. Anything above this is really unecessary, unless you are time stretching samples (think slow motion video, the more frames you have, the smoother it will be). When it comes to orchestral, I dont think time stretching samples is something many people are apt to do.

Posted on Wed, Sep 25 2019 21:22
by dmidi
Joined on Wed, Sep 18 2019, Posts 5

Originally Posted by: littlewierdo Go to Quoted Post

Not to mention the memory footprint. A full orchestral piece requires 60 plus gigabytes of ram for me, and that is really nothing in comparison with what it could be. There is a reason why 44.1 at a 16 bit depth rate was chosen for CD audio, this is considered just above the maximum quality that anyone can hear a difference. Anything above this is really unecessary, unless you are time stretching samples (think slow motion video, the more frames you have, the smoother it will be). When it comes to orchestral, I dont think time stretching samples is something many people are apt to do.

1. No disrespect, but have you read the thread?  This is about LIVE instrument playability quality, not RECORDED/MIDI music file playback sound quality or time stretching. 

To summarize, there is a negative effect of latency on pianist's sense of hearing and motor cortex processing as impacted by low sampling frequency.  That is, if you press a key and you expect to hear the sound in 3 Milliseconds, but the sound instead comes to you in 6 Milliseconds, that will be as if the sound board has moved from 3 feet away to 6 feet away.  A bit wierd to say the least.  And that assumes no other latency elsewhere in the sound processing chain.  As there always is this latency in every component, the cumulative effect of latency might mean that the piano sounds as if the sound board is 10 feet or more away.  And that's before we start to consider the effect of jitter, which can be described as the sound board constantly shifting back and forth several feet, with this perceived movement happening from note to note or even during the resonance of a single note.  Aye...very bad indeed for your brain to be 'hearing' the grand piano vibrating back and forth.

Also note that as this jitter is likely random, good luck trying to achieve consistent playback between pieces. And that is true even for playback of MIDI notes in which instruments are closely placed.

There is a way to elilminate the 'sound shifting':  By roughly doubling the frequency from 44 to 96 Khz, the latency of the VSL library is cut in half, and that allows for a more consistent and authentic playing experience. And by doubling again from 96 Khz to 192 Khz the latency is halved again and we have more 'latency headroom' to stay under the audibility threshold.

Don't take my word for it, but do take VSL's deafening silience on this topic as being problematic.  I suggest if my explanations are not sufficient, please research and understand other trusted sources.

2. As to RAM requirements, this piano at 24 bit 44.1 Khz currently has a RAM requirement of 4-6 GB.  Doubling the sampling frequency will half the latency and only require a total of 6 GB * (96/44.1) = 13 GB.  Most laptops with any sort of multimedia claims will have 16 GB RAM or more.

To editorialize a bit, VSL is doing an enormous dis-service to themselves and their customers by attempting to ignore the issue.  Unfortunately, an uneducated customer at some point becomes a frustrated customer...and that's a net negative for musical creativity and classical music as a whole (LONG LIVE POP MUSIC?).

Posted on Thu, Sep 26 2019 01:43
by robinlb1900
Joined on Sat, Jun 22 2019, Posts 17
I think VSL can provide at least two choices(44.1/96), let the users with different needs to choose.
Put aside the purely technical issues, just from marketing point of view,
this also shows VSL as a top sampling instrument supplier in world. I did it first:)
Posted on Thu, Oct 17 2019 04:15
by robinlb1900
Joined on Sat, Jun 22 2019, Posts 17
My friend got Newyork concert grand from Production voices, and they announced that the 96K version would be available soon. Size will be over 500GB by 9 Mics.
Maybe VSL can consider similar upgrade for users in future.
Posted on Thu, Oct 17 2019 10:28
by Dietz
Joined on Tue, Aug 06 2002, Vienna / Europe, Posts 7305

Originally Posted by: dmidi Go to Quoted Post
[...] By roughly doubling the frequency from 44 to 96 Khz, the latency of the VSL library is cut in half, and that allows for a more consistent and authentic playing experience. And by doubling again from 96 Khz to 192 Khz the latency is halved again and we have more 'latency headroom' to stay under the audibility threshold. [...]

Sorry to ask the obvious: Why don't you change your audio system's driver to those preferred sample rates if you want to bring down the latency? 

/Dietz - Vienna Symphonic Library
Posted on Mon, Oct 21 2019 12:14
by TFIS
Joined on Tue, Dec 14 2010, Posts 103

128 ≠ 128

Posted on Mon, Oct 21 2019 15:18
by Dietz
Joined on Tue, Aug 06 2002, Vienna / Europe, Posts 7305

Exactly, that's what I meant to say. If you switch the audio system's clock to 96 kHz, the latency will go down.

Kind regards,

/Dietz - Vienna Symphonic Library
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