Thanks all for the input. I think this thread settles it for me. It's not worth to upgrade to Pro for $300 just to get 192kHz for home usage where one just wants to play the piano live. I would rather spend the money on buying some instruments or for the upgrade to Pianoteq 7 (which should be this year as per the historical trend...)

Thanks again all.

Osho

You're wise.  The main benefits of Pro are in the control of the Presets rather than the increased sampling.

The only use case where I've encountered needing a higher internal sampling rate is if PTQ is being integrated live into broadcast or professional production pipeline hardware that has a specific inbound rate requirement either for convenience (like in a production pipeline where data is being shared in multiple workflows through multiple production iterations and shouldn't be getting dithered as its sent back and forth between technicians and contributors) or for live broadcast hardware requirements (where again dithering/resampling either may not be possible between inputs on some hardware or where it could create artefacts or other problems because of downstream mismatches).  While needing 192 is usually fairly rare as a requirement in any setting, I regularly have studios request 96 so they can downmix and dither at the end of their pipeline rather than try to mix in something that has to be upsampled before entering the pipeline.  As more and more is done in-the-box even for live or production workflows using circuitry that encodes and decodes using calculus instead of statistics (like delta-sigma DACs), the practical reasons for different sample rates will likely decrease significantly over the next ten years as more legacy digital hardware is upgraded with newer, faster, more flexible hardware--especially if the trend towards streaming instead of broadcast continues.

There are particular format reasons for why these sample rates are set to what they are (NTSC American audio formats linked to multichannel color at 44056, Audio CD max file rates at 44100 because of Sony's PAL/NTSC average, early ADCs and DACs at 48000 for higher-fidelity than CD, etc.), but any additional sampling above 50kHz doesn't improve the listeners sound of something, it just increases the headroom for the amount of FX, dubbing, processing, and downsampling a track can handle without the foldback artefacts mentioned above.  As delta-sigma converters and their pulse-density modulation have replaced most traditional pulse-code modulation in digital signal processing and CPUs and GPUs doing more and more heavy-lifting in digital audio production, this whole question could functionally disappear within a few years.  Already in most of today's hardware both consumer and professional, data compression ratios and codecs are the most important factors for sound quality, while concerns around sample rates and bit depth have evaporated.  Even network bandwidth and datastream size are functionally irreverent as 4K video formats/codecs/encodings are so many Mbps that it doesn't matter how large the audio packets are, because the video will always be so many magnitudes larger in the datastream nobody cares how much space the audio uses anymore--a far cry from the SD DVD days.  The problems with codec containers and bit rate get even worse when trying to compare any shared examples because nearly any website that accepts uploads will re-encode, compress, and resample them anyway.

My experience is that if you need this feature someone else will tell you as part of any agreement to contribute to a workflow pipeline, but if you're working on your own or just want to play the piano, the difference is practically non-existent.

Further reading on sample rates and delta-sigma DAC algorithms for anyone interested:
https://en.wikipedia.org/wiki/Sampling_...o_sampling
https://en.wikipedia.org/wiki/Delta-sigma_modulation

Thanks again all.

Osho

I was just playing some piano music!

Here’s some music; as I play it I can of course just faintly hear a piano part.

While under your Perf tab you can downsample, any upsampling is yet unsupported.  However, on my copy of PIANOTEQ PRO, it permits a 192kHz sample rate in file exports (although my audio interface maximum is 96kHz).

Thank you Viridis, for your prompt reply.

Please correct me if I'm wrong; 48kHz is a new Sample rate limitation of both PIANOTEQ Stage and Standard.  I take it if I am right end users essentially accepted a downgrade and without so much as a single protest!
Nobody appears at all concerned about an omission in the change log

Internal sample rate of 48 kHz has always been the maximum for stage and standard.

No change log indicates an adaption of a 48kHz ceiling under Options.  Please correct me if I'm wrong; 48kHz is a new Sample rate limitation of both PIANOTEQ Stage and Standard.  I take it if I am right end users essentially accepted a downgrade and without so much as a single protest!

Nobody appears at all concerned about an omission in the change log.

EDIT: I left off a period: picked up a bad habit.  Have to steer clear of bad influences.

Anyway, in any event I'm hardly misled by your assertions!

Oh, another word about sample rates, somebody confuses them

That is what I referred to as "false" - I believes ya knows that - don't ya buddy?

In the past I posted about dithering. For one example, here's what Ra said in one thread:

.

His own words show, I brought these ideas to him.

Link to that quoted thread here.

Here's a pertinent part where I informed him of these notions around dithering.

At that time, I was happy to have shared this knowledge which might help him in his ongoing learning.

If I ignore the insult and all, here's more info..

I recommend to watch the whole video posted by Chopin87 at top.. at around, or maybe before minute 10, or from the start really, it's worth it.. he's absolutely nailing a presentation of the way that working with frequencies in a DAW is different at the different sample rates.

All I tried to add to the understanding here is that, depending on output which might include dithering options, a downsampled 44.1kHz version of something which was once heard at 192kHz could be very different because of all the technical choices during downsampling.. so comparisons are quite often not 1:1 - just an observation.

Absolutley related things (inharmonicity being high frequency related) - like I suggest Ra, go to that video, and see how the demo with sine waves sounds (he shows these things beautifully and it's why I mentioned 'synths' lot of high freq's. - the simple wav forms coupled with these artifices esp. in high frequencies, can be powerful components in your sound design, more than just layering a track over another.. these things can be complex quilts woven quite intricately - and playing with the technical areas of it can lead to some artistic satisfaction - all I'm trying to gift people with here).. any high frequencies exactly relate to my earlier postings. Inharmonicity may show up in high frequencies and be subject to some intermodulative processing.

Like scherbakov.al mentioned, you may hear a sweeter 'tail' to chords etc. at 192 - one reason being exactly what I was hoping to express.. these 'artifacts' of intermodulation can make 'pure' waves noisy ast lower sample rates.. again it's both good and bad though.. and I pointed out, I like 48.. and Chopin87's video made my day - I didn't know before this, that 48 has such a nice technical situation which applies to my desires.

So, working on piano at 192 is good.. I hinted at that earlier..

With different sample rates in the DAW, all kinds of frequency effects run rampant esp. with effects on top. It may be different for given instruments, effects, the genre - but your choices can help or hurt the outcome.

You choose the rates you want to work in, partly because of what you want to either keep, or destroy in the working digital signal. You can create 'tape' vibe with some simple destructive digital planning and I love going 'glitch' to degrees.

Next, you mix - whatever other production you want to apply (again, you can choose certain compression types, which variously help to keep or destroy values in the signal)..

Next, if you want to output to a consumer sample rate 48 for example, you might apply at that last stage, any given type of dithering, or not.

In the case of keeping a cleaner output to a non-lossy format (like flac or WAV) - you may avoid dithering - but a lot of the time, the process of downsampling really does leave some harsh areas, masking problems - so. This is one reason dithering is done.. to help the human ear - the translation downward is pertinent to what we hear.

If outputting to and MP3, the MP3 format's own compression routines may cause some negative consequences for the downsampling too.. so you may not want to dither. Again, variables incl. different music types, what rates it was 'worked on' and more, just an example.

I hope you understand I'm trying pretty hard (over a long time line) to give information you can take onboard.. not trying to just "say stuff" - and please quit with the insinuations OK?

In the end.. it does depend on hearing the same thing on the same equipment producing the same level of authenticity to the signal, otherwise we can be comparing apples to oranges - just hoping to shine that light on it.

Sincerely - just hoping this helps people - not trying to 'win' stupid prizes mmmmkay

Honestly, I was unaware you could just use 192kHz as an Internal sample rate even though your audio interface might max out at 96kHz like mine!

Thank you, Osho.

And yes if I do say so now myself, differences are subtle.

scherbakov.al wrote:

The difference was found in the long decay of complex polyphonic chords. I found more naturalness in sound space. But this difference is subtle. (After 20-30 seconds of sustain, differences can be discerned). It's not about the presence of high frequencies, but a 4-times more accurate calculation of the available frequencies.

Have you verified ‘a more accurate calculation’ via a spectrogram?  One which shows comparative differences in a long decay, and in some histrionic.

You can listen to an example:

44100

192000

Converting to MP3 retained some of the difference. Only space has become flatter.

An example from 2017. Since then newer versions of Pianoteq have been released, maybe now there will be no difference .. I do not know. Did not check again.

The difference was found in the long decay of complex polyphonic chords. I found more naturalness in sound space. But this difference is subtle. (After 20-30 seconds of sustain, differences can be discerned). It's not about the presence of high frequencies, but a 4-times more accurate calculation of the available frequencies.

Have you verified ‘a more accurate calculation’ via a spectrogram?  One which shows comparative differences in a long decay, and in some histrionic.

Also, what is the relation between sample rate and audio buffer size?

192kHz has sometimes less latency than 96kHz.  At 192kHz a buffer of 32 samples happens twice as fast as it does at 96kHz.

If you’re proposing 192,000 samples per second makes an audible difference compared to 96,000 samples in the same time frame, you present no evidence supporting such a claim.

You may look at internal sample rates as computational speed occurring inside the software itself and device sample rates as any audio interface device or soundcard settings.  I know I do.

When I change the Sample rate under the Devices tab, it immediately changes on my audio interface.  From the software it affects the audio interface.  Conversely, whenever I change the Internal sample rate under the Perf tab, my connected interface is unchanged.  Internal sample rate as it implies is internal; it is inside the software.  It has no affect on the hardware audio interface.  Which is an external hardware device.