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Been a while since the last one, and these are quite fun:

https://www.dropbox.com/s/xk2n3bm95lzm6fg/Prelude%20-%20take%201.wav?dl=0

https://www.dropbox.com/s/oo810m2bea28ceh/Prelude%20-%20take%202.wav?dl=0

https://www.dropbox.com/s/3ey6s0hfql9tg15/Prelude%20-%20take%203.wav?dl=0

Let's see who can tell what these are (looking at you CyberGene )...all three files were played with their respective VST/Piano etc. and no changes were made (bar from adding some built-in reverb).

I'll give it a day or two and then will publish answers..

Cheers,

J

Number 3 is pianoteq. Can’t tell the first two

To me all 3 are different presets of Pianoteq sounds...

Almost missed this thread. Not sure it’s the right piece for demos since there are only short notes while longer sustain would have been better as well as varying texture rather than almost monophonic arpeggios. And we need more octaves...

Anyway, to me 1 sounds best, 2 is so-so, 3 is ugly. I like how 1 sounds balanced with no exaggerated dynamics. 2 and 3 suffer from some harshness where louder notes stick out like they are artificially amplified selectively. And the general timbre of 3 is kind of boxy. Hard to say what is what from this type of demo but if I have to guess, 3 is Pianoteq.

The three are distinctly different ... but CG is right: the tune being played doesn't reveal much. (Was that even real music?)

Resonance sounds great on #2 to me. #1 feels like a tone generator and #3 is terrible, whatever it is. I do hope #3 is PTQ, though. I wouldn't be surprised if all 3 were PTQ.

1 and 3 sound the same to me. Both sound pretty flat. 3 is just louder and might have some sustain/reverb?

2 has a lot more dynamics. Still not the best piano sound I've ever heard but better than 1 and 3 to me.

#1 Sounds fine, not Casio or sample based Roland
#2 Help! My head is stuck inside a piano! Is this is a "loudness war" version?
#3 Same as #2

They all have a very similar timbre.

#1: Brittle sound, and very artificial
#2: Much more realistic, but a rather hard sound
#3: Intermediate between the other two in every respect.

Personally, I would prefer #2. It would drive me to insanity, if I were forced to use #1 for longer periods of time.

I wrote this last night, listening to the files through my AirPods. I just re-listened to them through my HD-650 and while my comment still holds true, there's this noticeable difference in volume levels. You should have normalized all files. 2 and 3 are too loud compared to 1.

I think the 3 is even nastier through proper headphones. 1 is a slight bit too clean and dry but it seems to respond best to dynamics and the soft timbre is most to my taste. 2 is not bad in terms of timbre but there's something wrong with its dynamic response, it's too sensitive and jumps out unnecessarily loudly and that ruins it. Also I think the reverb in 2 is not very well done. Judging from the proper dynamic response in 1 and very unnatural dynamic response in 2 and 3 I would assume 1 is your digital piano's internal voice, whereas 2 and 3 are VST-s and you need to rethink your velocity touch settings.

Kudos CG - I don't know how you do it but you do have a knack picking these out...

1) was built-in CFX sound of P515 (played on P515) - with edited paramaters: Concert Hall/Reverb 15 - recorded to USB (and transferred to dropbox)

2) was VI Labs Ravenscroft 275 (played on SL88 Grand) - with TW Hall Warm/Reverb 20 - recorded using GarageBand DAW (no other added effects bar a little compression)

3) was Steingraeber E-272 using the Venue Reverb from the Ant. Petrof Venue preset - recorded directly to PTQ (end exported as WAV)

I have learned a little something from this, the CFX with the P515 is probably the best combo I have, in terns of sound/action/eveness ratio to the SL88 with the VSTs but I still don't understand why the P515s USB recording does not produce a decent output volume, when I play it over speakers vs hearing it back is lacklustre in comparison and just doesn't do the playing any justice whatsoever.

Quick question, is the answer to normalise the recordings and if so does that not affect the dynamics of the performance or should the gain be used instead? Other recordings of the P515 especially on YouTube are so much louder with clarity, I know there is something missing...

I don't know what specifically "gain" means here, but in normalization the dynamics will generally be increased. Imagine the sound level originally only reaches half of what is possible. Normalization will then double the sound values, so the difference between no sound at all, and maximum sound level, will double. This increase in dynamics implies a loss of sound quality, because digital sound moves in steps, and in this present scenario the steps will become twice as large. Whether this loss of quality is truly audible is far from certain.

You should definitely use normalizing, it's a non-destructive process that doesn't change anything (well, there is rounding, but anyway) besides detecting the peak value and then amplifying the entire file so that the peak value hits a target, e.g. the max value without clipping which in the digital domain is 0dB. It doesn't change dynamics because the entire file gets equal digital gain. BTW, when exporting from GarageBand, the files are by default normalized to 0db (or maybe -1dB, that doesn't matter) which is why 2 and 3 are already normalized. Normalization is a two-step process: first you go through the entire file to detect peak velocity and thus calculate the gain and then apply the gain. Which is why digital pianos can't do that because they stay on the safe side and record relatively quiet on the USB file and don't know how loud you would play. After you stop playing, all is finished and I guess it would be a bit complicated for the piano to re-read the file from the USB, detect peak and normalize. On Kawai pianos you can apply a fixed gain to counteract the usually quiet USB volume but that's not wise because it would lead to clipping and that's already a non-recoverable distortion.

Long story short: after you record to USB, just open the file in Audacity and apply a normalization to -1dB which is a pretty standard procedure. (Don't mistake it for dynamic compression which applies different gain to different parts of the file to make it sound always loud.)

No, that's not true. Dynamic range is defined as the difference between the loudest and quietest. Imagine that in a file the loudest part is -30dB and the quietest is -40dB. The dynamic range is 10dB. What normalization will do is detect the peak value of -30dB, and if the target value is -1dB will apply 29dB gain to the entire file, so the new peak value will be -1dB and the new quietest value will be -11dB, so the dynamic range is again 10dB, it's not changed.

Thank you CG - this does make a huge difference! Just normalised a file in Audacity and this brings the performance that much closer to how I hear it when I play - examples of pre and post:

Before normalisation:
https://www.dropbox.com/s/mfxci28dpdc5nmp/BLUESINTHEATTIC-P515old.WAV?dl=0

After normalisation:
https://www.dropbox.com/s/9pl15ceqzfhq4la/BLUESINTHEATTIC-P515.wav?dl=0

I am liking the CFX even more now! :-)

I get your point entirely, but you're forgetting that the quietest level is zero, and when zero is multiplied by anything, it still is zero. Therefore any change in the maximum sound pressure level must necessarily alter the dynamics.

I just noticed that we're not talking about exactly the same thing. You made specific reference to the concept "dynamic range", whereas I am talking about dynamics quite generally. That may be the cause of the difference in our definitions.

Yeah, there's some misunderstanding here. There's always a noise floor. Which is why dynamic range is defined as the difference between the loudest value and the quietest value (i.e. the noise floor) and since you also apply gain to the noise floor, the dynamic range is the same. As to dynamics... well, depends on how you define dynamics. But when you normalize you also increase the noise floor, so SNR stays the same. Remember there's no such thing as "zero". There's a level below which you can't discern between signal and noise. Even digital files have noise floor determined by bit-depth, 6dB per bit, so -96dB noise floor for 16-bit audio.

This is definitely a good idea in general. In this case, where comparison of recordings is the goal, it would even be better to normalize based on a statistic of the average value in the recording, e.g. rms amplitude, instead of the most extreme value. This is more complicated because you need the determine the rms amplitude of each recording, and find a set of scaling scaling factors that equalizes rms across recordings but that at the same time do not lead to samples being larger than 0dB in any recording.

Are there any tools/apps that do this automatically across a given set of recordings? If not, then there might be an opportunity for someone to make an app. A slight enhancement would be to use the rms amplitude of each recording to to the same level, but where as a group, they are as high as they can go with none of the recording in the given set of recordings, clipping.