[linux-audio-dev] RE: disk-based real-time sampler prototype released ! please try it.

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Subject: [linux-audio-dev] RE: disk-based real-time sampler prototype released ! please try it.
From: Garth Brantley (garth_AT_fullduplex.com)
Date: Thu Jul 13 2000 - 20:28:20 EEST


> -----Original Message-----
> From: Benno Senoner [mailto:sbenno_AT_gardena.net]
> Sent: Thursday, July 13, 2000 1:27 PM
> To: Garth Brantley
> Cc: linux-audio-dev_AT_ginette.musique.umontreal.ca
> Subject: RE: disk-based real-time sampler prototype released ! please
> try it.
>
>
> On Thu, 13 Jul 2000, Garth Brantley wrote:
> > Awesome man! I cant wait to get home and get this loaded up.
>
> hehe, thx, don't forget to generate the sampler with a bit of
> vibrato so
> that they sound less static. :-)
>
> > Once Richard gets his feature list up I will go over and
> add some suggestions.
> > I've spent a decent amount of time in my life playing with
> various hardware
> > & software samplers so this should help. As far as the DSP
> programming
> > stuff: What is the current state of plug-in APIs on Linux?
>
> LADSPA 1.0 specs are complete and are suitable for pure FX processing
> (like Reverb Hall filters etc).
> There seems already a good ,free opensource reverb available:
> Jezar's Freeverb at http://www.dreampoint.co.uk
> (BTW: has anyone tried it ? is it really as good as stated on
> his page)
>
>
> > A successful sampler needs excellent filters & DSP modules
> such as bass
> boost, comb
> > filters, etc. But as far as effects like reverb, delay,
> chorus, etc,
> > plug-in support would be ideal.
>
> yes especially because the you can use one single reverb for
> the whole
> instrument/sampler by using the FX send amounts.
>
> Regarding filters (like resonant LP for example), I am not sure for
> performance reasons, it it would be better to inline them
> instread of calling
> plugin functions. But if the plugin approach works here too,
> then it would be
> kinda cool to have tons of interchangeable filter for each
> instrument/voice.
>
    Well traditionally with hardware samplers & with gigasampler, the
filters are built in. They need to be applied to separate layers of the
sound separately & are a fundamental part of what a particular program
sounds like.

>
> > As far as the file format thing goes. I don't think
> any of the current
> > formats out there (Akai .wav & .akp, Sound font, .krz, let
> me know if I am
> > missing any) are adequate for a comprehensive pro sampling system.
> > I think the file format would be an ideal place to try to
> get some of the
> > existing sound designers & industry people into the
> picture. There is no
> >question that the sound design industry and consequently the
> musicians who
> > buy samples have a lot to gain from an open cross-platform
> sample library
> > format. The biggest hassle in putting out a sound library
> is dealing with
> > all the different incompatible samplers out there.
> > Anyway I am still doing research on this & I will post
> something here to the list by Monday. If
> > anybody knows anything at all about the .Gig format please
> let me know.
>
> I fully agree on the need of the open format but when
> designing we need people
> which know the problematics of the realworld.
> Speaking of myself I have only used "toy" expanders like the
> Roland Sound Canvas. Although I played around with a sampler
> of a friend of
> mine (long time ago) , I have not a very clear picture how
> the layering works.
> ( at least from reading in magazines Roland's JV1080 series
> seem to have a nice
> voice structure, plus they sound really good)
> So someone which could enlighten us about what is exactly
> needed to satisfy the
> professional users, would be very useful.
>
    Well I have worked with Akai, Kurzweil & Gigasampler. With each system,
one program can have a fixed number of Layers. A layer is defined as one
sample voice & associated envelopes & filters. On akai S1000/s3000 you
could have only 3 layers/program. On Kurzweil & giga you can have a lot
more. The layer system was create to have different samples for different
velocities. IE you record a piano key being struck light, medium & hard.
If the keyboard is struck with a velocity between 0-40 you play the light
sample, between 41-80 the medium sample, and between 81-127 you play the
hard sample. Most samplers seem to provide a method for crossfading
velocity layers. IE the light sample runs from velocity 0-50 and the medium
sample plays from velocity 30-90. Between velocities 30-50 the light &
medium samples are crossfaded.

> Regarding the samplelibraries:
>
> I think audio companies have a very closed attitude (see
> fileformats),
> and often the sample-library producers have deals with
> certain synth/sampler
> manufacturers which basically ties the samplelibrary to
> determined sampler
> models.
> For example let's say firm XYZ produced a cool samplelibrary
> in .GIG format,
> and you ask them to port it to an open sound format ?
> Since they have probably a crossdeal with the GIG folks ,
> they will be reclutant
> to port it , because they worry about lost GIG SW sales.
> (I may be wrong but that is my impression)
>
> A reverse-engineering of the GIG format would solve this
> problem, but of course
> is not an ideal situation. ( they could keep the format
> changing, plus you have
> not full insight into the details of how filters/FXes work etc).
>
    So far the .Gig format seems like a big pain in the ass to reverse
engineer! It doesnt seem anybody out there has even really tried so far.
And a quick look at a few .GIG in a binary editor doesn't give me much
confidence that it would be easy to do.

> > ps - Once I get something together, I am going to make a
> posting to the Akai
> > & Kurzweil sampler mailing lists. I know a few
> professionals on these lists
> > that have the potential to get their companies behind some
> sort of open
> > standard file format.
>
> keep us informed.
>
> Benno.
>

 -Garth


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