Posted by: Toby

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Sure there is if you use the hierarchy the way it was designed. The doc shows the hierarchy

A Yamaha 'waveform' consists of up to 256 key banks with each key bank containing one mono or stereo sample (e.g. external wav file with loop points if needed).

For each key bank you can specify the note range and the velocity range. This allows you to create layers, even overlapping layers. The sample selected will depend on the actual key you press as well as the velocity you press it with.

Yes, but this will be one shot sample, no loop edited and a very loooong job too

Posted by: Toby

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Not quite correct. You do NOT assign keybanks to elements - you assign a single Yamaha 'waveform' to an element. That waveform can contain multiple keybanks, note ranges and velocity ranges as mentioned above.

Of coarse it is what i say, you cannot use add keybank fron editwaveform to import a multiple edited keybank.

. This is not a way for me, it need to import waveform, not wave. 

Posted by: Toby

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Sure there is if you use the hierarchy the way it was designed. The doc shows the hierarchy

A Yamaha 'waveform' consists of up to 256 key banks with each key bank containing one mono or stereo sample (e.g. external wav file with loop points if needed).

For each key bank you can specify the note range and the velocity range. This allows you to create layers, even overlapping layers. The sample selected will depend on the actual key you press as well as the velocity you press it with.

Yes, but this will be one single shot sample, one key and not loop edited, add sample bring to only audio file witch I suppose can not be looped, am I wrong?

Posted by: Toby

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Not quite correct. You do NOT assign keybanks to elements - you assign a single Yamaha 'waveform' to an element. That waveform can contain multiple keybanks, note ranges and velocity ranges as mentioned above.

 it is what i wrote , you cannot use add keybank fron editwaveform to import a multiple edited keybank.

From the User Waveform menu, press the "Add Keybank" button.

Since Montage 1st gen and Montage M both removed the sampler features from Motif XF the new workflow is to add loop points with external software.  You also have to normalize and otherwise prepare the audio files with an external device (sampler, computer, etc) before importing.

I wouldn't mind having some basic sample editing features even if not full a full blown sampler feature set.  But I think I would probably prefer development to focus on other areas of the keyboard instead.   

For example:

I would prefer to have imported audio files to be internally compressed so memory consumed by user samples is reduced.  Saving user waveforms to files (user/backup/etc) could save user waveforms uncompressed in the file if Yamaha is worried about reverse engineering the compression algorithm.

Posted by: Jason

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add loop points with external software.

I did not know that a wave file could be embedded with a loop point, I though only waveform could do that. But the workflow thru add keybanks is a too long process for built velocity multiple layers .

But how that work for avalaible awm voices in montage? I means, 

— building in one element thru add keybanks, with let say 4 velocity layers made by 4 multisample of 10 keybanks each, that’s made 40 keybanks of 4 different instrument.

versus

—using 4 element thru loading waveform, each element have 10 keybanks of one instrument

is the number of voices used is the same in both case? And how many?

Assembling user waveforms is a chore since I agree the process is clumsy.

Things may be improved if they would let filenames designate root note and other information then allow loading more than one file at once.

Even better to allow loading some standard multi sample format.

For now there's a lot of steps.

Melas tools also can help.

So it seems that it is one voice per element, one part of 8 element use 8 voices and one performance or 8 part by 8 element will use 68 voice for one key, that means you have a two notes polyphonic keyboard. If you could made a 4 layer multisample per element using add keybanks, you could multiply by 4 the polyphony, I know it’s a kind of absurd to use at max all part but it give me the limit of the engine. But If you put more layer in one element, that is mean I suppose you will lose definition of the sound or not? 

So it seems that it is one voice per element, one part of 8 element use 8 voices and one performance or 8 part by 8 element will use 68 voice for one key, that means you have a two notes polyphonic keyboard.

That pretty much sums it up - except I would caution you to NOT use the word 'voice' since that word has a different, and specific, meaning for other keyboards.

You didn't mention parts 9-16 that could also be sounding and being being controlled by an external source.

Then there could also be more than one arpeggio active and they can play repeatedly even if the user isn't playing.

And, if that isn't enough to worry about, you must also consider the sustain pedal. It can keep old sounds going, and using polyphony, as the user plays new sounds.

If you could made a 4 layer multisample per element using add keybanks, you could multiply by 4 the polyphony,

That won't work - there are only a limited number of processes that can create sound. It doesn't matter if you use 4 at a time or 1 at a time - the limit of 128 preset and 128 user is still the same.

I know it’s a kind of absurd to use at max all part but it give me the limit of the engine.

Maybe - but the point you raised is valid for perfectly normal use. See my tests/examples in the 'Voices cutting off in Montage' thread.

That user is asking about why, in the example video they provide, the sound is cutting off.

Take your numbers above and consider a user playing multi-note, two-handed chords, as might be common when playing gospel music, and also using the sustain pedal.

3 notes per hand -> 6 notes

4 part performance - part 1 has 5 elements, each of the other 3 have 2 elements each -> 11 elements

If each of the 11 elements sounds for each of the 6 notes that uses 66 notes of polyphony - just over half the total.

Press the sustain pedal and play another, similar, two-handed chord and you add another 66 which exceeds the limit.

It isn't as absurd as it might seem at first. You DO have to be careful to make sure the performance/part/element setup matches what, and how, you intend to play.

The "best" usage of all of these extra elements per Parts would be to increase either the velocity density, cycle density, or use of [ASSIGN 1&2] buttons for switching between different sets of elements to "replace" using Parts to switch between sounds.

None of these techniques consume more polyphony since the idea is that every piano key strike only initiates one sample.

If XA control can add some more controls (super-duper articulation) then these could also enhance the utility of all of these extra elements.

The "best" usage of all of these extra elements per Parts would be to increase either the velocity density, cycle density,

As an experienced player do you have any sense as to how accurate a player can be in consistently hitting a specified velocity range?

Automation, or an arpeggio, can hit any velocity repeatedly and consistently. So for automated/background purposes that is great.

But what might be typical human limits?

The 2.0 Data List, and tests, show all of the velocity ranges are still 0-127 (e.g. not extended to 1023).

But can a human consistently play notes in a range that is 5 wide? An even better question is can a player even remember they need to do that for certain measures.

If the velocity mapping changes throughout a tune (perhaps different instruments or articulations are involved) how would a human even remember they need to hit a particular velocity to get it to sound the way it was designed?

Part of the appeal of synthesized sounds like FM-X or AN-X was how "expressive" they are and a lot of this is due to the variation in sound from velocity.  There's 127 different sounds possible in terms of "character" of the sound and although you can't really command your hand to hit all 127 variations, hearing more than 8 "steps" of variations is absolutely achieved.  So if you can make AWM2 have higher velocity resolution where there are subtle timbral differences so you somewhat approach a modeled sound's capacity to algorithmically arrive at timbral variation has its utility.   Will you or anyone else be able to force out all variations at will?  No.   Will higher velocity resolution be noticeable by the average player?  I would argue yes.  I don't even think you "need" higher velocity resolution offered by MIDI 2.0 to take advantage of this.

I think the bigger wall to run into is sample memory, compression ratios, etc.   It's expensive (memory footprint) to compete with modeled sounds using samples.   Therefore, not many sounds can take advantage of this without filling up available memory quickly.

This goes for cycle chains too.

The XA use where you switch out elements in one Part to create multiple instruments in one Part doesn't cost as much because the thought is to basically stick with 8ish elements per sub-part-instrument (using the same built in waveforms as now) but have "a bunch" of them.   Trading off flexibility in the Part level settings that would otherwise be different if you split the same sounds into Parts.

Posted by: Toby

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The "best" usage of all of these extra elements per Parts would be to increase either the velocity density, cycle density,

As an experienced player do you have any sense as to how accurate a player can be in consistently hitting a specified velocity range?

Automation, or an arpeggio, can hit any velocity repeatedly and consistently. So for automated/background purposes that is great.

But what might be typical human limits?

The 2.0 Data List, and tests, show all of the velocity ranges are still 0-127 (e.g. not extended to 1023).

But can a human consistently play notes in a range that is 5 wide? An even better question is can a player even remember they need to do that for certain measures.

If the velocity mapping changes throughout a tune (perhaps different instruments or articulations are involved) how would a human even remember they need to hit a particular velocity to get it to sound the way it was designed?

 

 

 

 

that are the limits of velocity triggered by fingers, that’s make a huge gap when using choirs, strings or brass orchestra, and even if velocity, volume or whatever is triggered by a foot pedal, you need post editing of the recorded midi curves for Reach a ultra realistic orchestra score sound. 

Posted by: Jason

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 Therefore, not many sounds can take advantage of this without filling up available memory quickly.

That is the point! It is like having a Ferrari with an oil tank of 1 litter. And this is not a technical issue I guess, a space in the montage managed for insert a user SSD drive wouldn’t have been great? 

That is the point! It is like having a Ferrari with an oil tank of 1 litter.

I'm just guessing you meant 'gas tank' rather than oil tank.

But I see the point more as using the right tool for the job.

Don't drive your Ferrari to the super market. Take your electric car or the hybrid.

If you want presets that are designed to be played 'as is' buy a stage piano or arranger.

A synthesizer is designed so that YOU can program the sounds you want. The presets provided are designed to demonstrate what it is possible to do on the instrument. Those presets are NOT necessarily designed to be used 'as is' for all types of performance use cases.

Sure you can pound a nail with your shoe but a hammer is the better tool for the job. Use the right tool and you will get better results.

Posted by: Toby

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That is the point! It is like having a Ferrari with an oil tank of 1 litter.

I'm just guessing you meant 'gas tank' rather than oil tank.

But I see the point more as using the right tool for the job.

Don't drive your Ferrari to the super market. Take your electric car or the hybrid.

If you want presets that are designed to be played 'as is' buy a stage piano or arranger.

A synthesizer is designed so that YOU can program the sounds you want. The presets provided are designed to demonstrate what it is possible to do on the instrument. Those presets are NOT necessarily designed to be used 'as is' for all types of performance use cases.

Sure you can pound a nail with your shoe but a hammer is the better tool for the job. Use the right tool and you will get better results.

 

 

don’t agree at all, you are missing the point so evident. Montage is a wonderful instrument made for you can programmed your own sound and bring them to the stage and then you need memory for that. Montage have now the capacity to put at the tip of your finger( specially because of hight definition and 128 element per part) orchestra , strings or choir that can challenge soft dedicated for scoring music like in kontakt engine.  But it miss a simple thing, memory. A choir patch or orchestra can need much more than 10 gb. I use my montage classic as it is , I enjoy it so much, and even I don’t use all memory sample for the next spectacle. I know the limitation and I don’t try to put articulated choir or string library on it. But if I should invest in the new montage and it should be worth it, I will feel much more this contradictory fact to have a Ferrari that you can feed only with 1 litter of gas.

And if I was one of the creators/ engineer of the montage that make a so great job, given (selling should be the word) to us the best synthesizer  I could find in the market for my needs, I will feel even much more this contradictory hardware building, what are technical issues for an increase memory apart from commercial, business point of view? But this is philosophical discussion and I will not further as this world is full of contradiction coming from divergent interest and it always been like that. More I analyze the market, more I find these kind of contradictions and I can say because of my analyses that montage and montage M are miracle, believe me.