redhexagonal

i'm trying to integrate a sample player built in maxmsp with a rewire host ,usually ableton.

how do monophonic samplers manage to play a loop triggered by midi, without clicking? (since they cant know in advance when the next midi note will occur)

i'm trying to integrate a sample player built in maxmsp with a rewire host ,usually ableton.
how do monophonic samplers manage to play a loop triggered by midi, without clicking? (since they cant know in advance when the next midi note will occur)


how-do-samplers-work

adsr~ doesnt seem to work for me in this case. i dont see how i can fade to zero gain *before* retriggering the sample ,other than having every single midi note made a little bit short (a pain in the ass) to allow the envelope to close in time befor i retrigger it.

i was wondering how a commercial sampler app deals with this problem

it could be very great and interesting if somebody could answer to this question.

and it's certainly very simple for you, o masters of max.

ps: i have only a solution for simple wave (thx ej):

it could be very great and interesting if somebody could answer to this question.
it's really not easy for young maxers...

and it's certainly very simple for you, o masters of max.
:)

Here is a solution taking advantage of info~ to know the length of the sample. If you trigger the loop, the envelope will automatically go to 0. then rise for the fade in/out time (default 10ms). The sample will play and then the envelope will fade out to 0 during the last 10 ms of the sample. Again, if you re-trigger the 

sample at any time, the envelope will re-trigger as well.

I put in a quick little loop thing as well, so you can see how this would allow any loop to continue without clicks at the start/end points (if thats something you wanted).

I've always found line~ and curve~ to be the best envelope tools, they do exactly what you tell them.

Here is a solution taking advantage of info~ to know the length of the sample. If you trigger the loop, the envelope will automatically go to 0. then rise for the fade in/out time (default 10ms). The sample will play and then the envelope will fade out to 0 during the last 10 ms of the sample. Again, if you re-trigger the 
sample at any time, the envelope will re-trigger as well.  

I put in a quick little loop thing as well, so you can see how this would allow any loop to continue without clicks at the start/end points (if thats something you wanted). 

I've always found line~ and curve~ to be the best envelope tools, they do exactly what you tell them. 

After re-reading your post, i think I may have misunderstood your problem. Here is another solution using two line~ objects to do the following things:

This way you can re-trigger the sample over and over again, click free. I tried it and the 10 ms total fade-in out time doesn't seem too bad, though you could tweak these values up or down to taste.

fade out the sample over 5 ms
(re)trigger the sample
fade the ssample back in over 5 ms 

so basically youre suggesting introducing a latency into the sampler to allow time to fade out. is this what the exs24 or kontakt or whatever does? how come you cant hear a latency in those instruments?

i would guess this concept isnt going to get on too well with a max sampler using poly~ thispoly~ etc.....

>adsr~ @legato 1? It won't force to start at 0 (like the [route >list] trick).

surely the envelope has to go to zero to stop the sample clicking at the start ....theres a discontinuity when the sample is retriggered, not caused by the gain envelope

i'm really sorry but none solution work well for me.

tried with "cello-f2.aif" in the examples/sound folder.

i'm really sorry but none solution work well for me.
tried with "cello-f2.aif" in the examples/sound folder.
??


use adsr~ and send a retrigger amount of 10ms say. when the second outlet of adsr~ goes from 0 to 1 the sample is retriggered, so it will always trigger when the envelope is at zero. i think this will work inside poly~ too

of course then i have a latency of 10ms, but i guess i can compensate in my host somehow

i think i may have the answer!
use adsr~ and send a retrigger amount of 10ms say. when the second outlet of adsr~ goes from 0 to 1 the sample is retriggered, so it will always trigger when the envelope is at zero. i think this will work inside poly~ too

in fact i had a conflict between adsr~ (max release) and adsr~ Lyon potpourri... strange.

I beleive you can work with two play~ or groove~ or whatever. This wait you won't have the 10ms latency... All you need to do is alternate for one to the other. When you start reading a sound from one, you fade the other.

I beleive you can work with two play~ or groove~ or whatever.  This wait you won't have the 10ms latency... All you need to do is alternate for one to the other. When you start reading a sound from one, you fade the other.


actually this seems to work better.....the envelope is a simple Attack-decay with retrigger setting also

>I beleive you can work with two play~ or groove~ or whatever. This >wait you won't have the 10ms latency... All you need to do is >alternate for one to the other. When you start reading a sound >from one, you fade the other.

that would double your CPU load though, if only for a millisecond or so

Quote: bin wrote on Thu, 14 December 2006 04:04

----------------------------------------------------

> adsr~ doesnt seem to work for me in this case. i dont see how i can fade to zero gain *before* retriggering the sample ,other than having every single midi note made a little bit short (a pain in the ass) to allow the envelope to close in time befor i retrigger it.

> i was wondering how a commercial sampler app deals with this problem

realtime programs do not _need to know the next (midi) 

event _BEFORE it comes to perform an action on audio 

(like interpolating or crossfading but also think about

realtime systems like ableton live or the plug-in interfaces

(VST AU RTAS...) do the opposite: the events from the

sequencing part of the software are sent _earlier than you 

without that simple trick cubase or protools would not even 

be able to start 10 tracks of disk audio at the same moment

( =within 1 sample =eventually 1/192 milliseconds)

in the case of ableton live (i.e. or a similar solution within

MAXMSP) you could also perform a complex declicking or

crossfading DSP _offline as soon as the main speed in the

"song" is changed. for such a system, you would use 2 buffers

for every sample slot, so that the calculation of newly

processed files will be fast enough so that one does not 

lastly, if all what you are looking for is a quick hack to

remove the bad bad click in a loop, why not just fadout

the sample slot for 10-12 milliseconds; noone will notice 

or give up drumloops, they stopped beeing cool in 1996.

Quote: bin wrote on Thu, 14 December 2006 04:04
----------------------------------------------------
> adsr~ doesnt seem to work for me in this case. i dont see how i can fade to zero gain *before* retriggering the sample ,other than having every single midi note made a little bit short (a pain in the ass) to allow the envelope to close in time befor i retrigger it.
>
> i was wondering how a commercial sampler app deals with this problem
----------------------------------------------------

realtime programs do not _need to know the next (midi) 
event _BEFORE it comes to perform an action on audio 
(like interpolating or crossfading but also think about
_starting a new audio event.)

realtime systems like ableton live or the plug-in interfaces
(VST AU RTAS...) do the opposite: the events from the
sequencing part of the software are sent _earlier than you 
think/see/hear them.

without that simple trick cubase or protools would not even 
be able to start 10 tracks of disk audio at the same moment
( =within 1 sample =eventually 1/192 milliseconds)

in the case of ableton live (i.e. or a similar solution within
MAXMSP) you could also perform a complex declicking or
crossfading DSP _offline as soon as  the main speed in the
"song" is changed. for such a system, you would use 2 buffers
for every sample slot, so that the calculation of newly
processed files will be fast enough so that one does not 
notice the process.

lastly, if all what you are looking for is a quick hack to
remove the bad bad click in a loop, why not just fadout
the sample slot for 10-12 milliseconds; noone will notice 
that in a drumloop.

or give up drumloops, they stopped beeing cool in 1996.


check out the audiorate-sequencing-looping folder in the max examples folder or msp examples or something. Groove duck I think is the name of the patch, good way to duck in and out loops at the begin and ending curtousy of jhno.

I thought it was 93 when drum loops went out ;)

check out the audiorate-sequencing-looping folder in the max examples folder or msp examples or something.  Groove duck I think is the name of the patch, good way to duck in and out loops at the begin and ending curtousy of jhno.  

I thought it was 93 when drum loops went out ;)


I'm not trying to play a loop, cool/uncool drums or otherwise

youre not understanding my initial question! i cant use that groove duck patch because i'm not playing a loop. the end point is unknown by the max patch!

the problem is to do with retriggering a sample, or playing a sequence of samples in sucession with no clicks between them, triggered by midi input.....

adsr~ has a retrigger function, but it seems to be hard to synchronise the triggering of a sample with the zero point after the retrigger ramp down. this is not such an issue with synth patches since its less likely to be a discontinuity in the audio being enveloped

the last patch i posted was giving me good results last night if anyone is interested in this sort of thing....

I'm not trying to play a loop, cool/uncool drums or otherwise
youre not understanding my initial question! i cant use that groove duck patch because i'm not playing a loop. the end point is unknown by the max patch!

While the[makenote]'s third inlet is fed by the seventh outlet of an 

Inside your poly~ player, you may add a smoothing mechanism (using 

line~) to get rid of clicks when retriggering : buffer the message, wait 

f.e chanfrault | aka | personal computer music

> >>>>>> |sublime music for a desperate people|

> I'm not trying to play a loop, cool/uncool drums or otherwise

> youre not understanding my initial question! i cant use that groove duck patch because i'm not playing a loop. the end point is unknown by the max patch!

> the problem is to do with retriggering a sample, or playing a sequence of samples in sucession with no clicks between them, triggered by midi input.....

> adsr~ has a retrigger function, but it seems to be hard to synchronise the triggering of a sample with the zero point after the retrigger ramp down. this is not such an issue with synth patches since its less likely to be a discontinuity in the audio being enveloped

> the last patch i posted was giving me good results last night if anyone is interested in this sort of thing....

|
[unpack 0 0]
    |
    [sel 127]
    |
    (startloop)

While the[makenote]'s third inlet is fed by the seventh outlet of an 
[info~] object linked to your buffer~.
Inside your poly~ player, you may add a smoothing mechanism (using 
line~) to get rid of clicks when retriggering : buffer the message, wait 
for line to finish, output the message.

f.e chanfrault | aka | personal computer music
> >>>>>> http://www.personal-computer-music.com
> >>>>>> |sublime music for a desperate people|

bin ray wrote:
> I'm not trying to play a loop, cool/uncool drums or otherwise
> youre not understanding my initial question! i cant use that groove duck patch because i'm not playing a loop. the end point is unknown by the max patch!
>
> the problem is to do with retriggering a sample, or playing a sequence of samples in sucession with no clicks between them, triggered by midi input.....
>
> adsr~ has a retrigger function, but it seems to be hard to synchronise the triggering of a sample with the zero point after the retrigger ramp down. this is not such an issue with synth patches since its less likely to be a discontinuity in the audio being enveloped
>
> the last patch i posted was giving me good results last night if anyone is interested in this sort of thing....
>
>
>
>
>


im finding buffering the message to restart the sample alot less accurate than delaying the audio from the sample player (before enveloping it).

sorry i was confusing y'all. my loop is a midi loop , not an audio one. there is no looping audio

No accuracy loss at all, unless you made it the wrong way. Works with or 

> im finding buffering the message to restart the sample alot less accurate than delaying the audio from the sample player (before enveloping it). 

No accuracy loss at all, unless you made it the wrong way. Works with or 
without loop.

bin ray wrote:
> im finding buffering the message to restart the sample alot less accurate than delaying the audio from the sample player (before enveloping it). 
>
>


> sorry i shouldnt have said i was playing a loop at first, its just about triggering midi without clicks, in a loop or otherwise

bin ray wrote:
> sorry i shouldnt have said i was playing a loop at first, its just about triggering midi without clicks, in a loop or otherwise
>
>


Quote: bin wrote on Fri, 15 December 2006 01:57

lets say a sample _usually starts with some waveform

smoothly beginning at 0.0 - and if not, the click which

so the "click problem" you see is only at the end of 

the sample - where it is cut off at an unknown time and

a solution could be to let your sample fade out 20 ms

or so instead of cutting it off immediatley when you 

if you ue groove~ instead of buffer~ you do not even 

need 2 buffers to play a sample with such an overlap.

Quote: bin wrote on Fri, 15 December 2006 01:57
----------------------------------------------------
> I'm not trying to play a loop, cool/uncool drums or otherwise
> youre not understanding my initial question! i cant use that groove duck patch because i'm not playing a loop. the end point is unknown by the max patch!
>
> the problem is to do with retriggering a sample, or playing a sequence of samples in sucession with no clicks between them, triggered by midi input.....

lets say a sample _usually starts with some waveform
smoothly beginning at 0.0 - and if not, the click which
would occur if wanted.

so the "click problem" you see is only at the end of 
the sample - where it is cut off at an unknown time and
instantly plays again from the start.

a solution could be to let your sample fade out 20 ms
or so instead of cutting it off immediatley when you 
retrigger it.

if you ue groove~ instead of buffer~ you do not even 
need 2 buffers to play a sample with such an overlap.

> adsr~ doesnt seem to work for me in this case. i dont see how i can

> fade to zero gain *before* retriggering the sample....surely that the

You'd still need two voices for a mono sampler, just any new note will 

switch of the old voice with its envelope...

bin ray wrote:
> adsr~ doesnt seem to work for me in this case. i dont see how i can
> fade to zero gain *before* retriggering the sample....surely that the
> only way to have no click... 

You'd still need two voices for a mono sampler, just any new note will 
switch of the old voice with its envelope...

-- 
Stefan Tiedje------------x-------
--_____-----------|--------------
--(_|_ ----|-----|-----()-------
-- _|_)----|-----()--------------
----------()--------www.ccmix.com


> adsr~ @legato 1? It won't force to start at 0 (like the [route list] 

But if you jump at the same time to the beginning of the sample, it 

would create a click, it works with synths though...

Emmanuel Jourdan wrote:
> adsr~ @legato 1? It won't force to start at 0 (like the [route list]  
> trick).

But if you jump at the same time to the beginning of the sample, it 
would create a click, it works with synths though...

> that would double your CPU load though, if only for a millisecond or

for 10ms to be exact, and only if you put it into a properly setup poly~ 

- but if you're concerned about that, I'd recommend to sing, this will 

bin ray wrote:
> that would double your CPU load though, if only for a millisecond or
> so

for 10ms to be exact, and only if you put it into a properly setup poly~ 
- but if you're concerned about that, I'd recommend to sing, this will 
bring your CPU down to zero...

> or give up drumloops, they stopped beeing cool in 1996.

I thought it was 1986, but I guess I was ahead of time that time. 

(Timing has been confusing me always thats why I can't keep it...)

Roman Thilenius wrote:
> or give up drumloops, they stopped beeing cool in 1996.

I thought it was 1986, but I guess I was ahead of time that time. 
(Timing has been confusing me always thats why I can't keep it...)

>> that would double your CPU load though, if only for a >millisecond or

>for 10ms to be exact, and only if you put it into a properly setup >poly~ 

>- but if you're concerned about that, I'd recommend to sing, this >will 

do you not think it would be a problem to double CPU load for even 10 ms? it would certainly screw up the audio output if you went above 100 percent load....crossfading isnt an option for me since i am working at around 50 percent CPU load normally

>bin ray wrote:
>> that would double your CPU load though, if only for a >millisecond or
>> so

>for 10ms to be exact, and only if you put it into a properly setup >poly~ 
>- but if you're concerned about that, I'd recommend to sing, this >will 
>bring your CPU down to zero...

do you not think it would be a problem to double CPU load for even 10 ms? it would certainly screw up the audio output if you went above 100 percent load....crossfading isnt an option for me since i am working at around 50 percent CPU load normally


the last patch i posted using line~ seems to give me clickless playback using a host sequencer, or from any midi input. albeit with a latency of the same time as the retrigger time.

however it would be useful if anyone could tell me to use adsr~ to do the same thing seeing as thats what it is supposed to be for? and its nicer for muting and stuff.

please build me a patch! you clever people

the last patch i posted using line~ seems to give me clickless playback using a host sequencer, or from any midi input. albeit with a latency of the same time as the retrigger time.
however it would be useful if anyone could tell me to use adsr~ to do the same thing seeing as thats what it is supposed to be for? and its nicer for muting and stuff.
please build me a patch! you clever people


Quote: bin wrote on Thu, 14 December 2006 19:28

> so basically youre suggesting introducing a latency into the sampler to allow time to fade out. is this what the exs24 or kontakt or whatever does? how come you cant hear a latency in those instruments?

I assume these samplers are all polyphonic, which means they will always start a newly triggered sample on a new voice (just like poly~), fading out the previous sample independently from the new one in its own release phase. This is a feature that of course uses more cpu power than a monophonic approach. If there would be only one voice I assume you'd get clicks.

Quote: bin wrote on Thu, 14 December 2006 19:28
----------------------------------------------------
> so basically youre suggesting introducing a latency into the sampler to allow time to fade out. is this what the exs24 or kontakt or whatever does? how come you cant hear a latency in those instruments?
>

> please build me a patch! you clever people

I didn't give an answer to your last post because of this. Don't order 

people to 'show a patch' as i usually do when asking for C code 

1 - you don't explain your problem _clearly

2 - you're way too much nervous & aggressive to give people the taste to 

Again, the 10ms solution is the best one in any case, mono or poly. 

Human usually detect a delay starting at 20ms... I use this solution 

since years, in *every* MSP patch i have, to avoid any kinds of clicks. 

5ms fade out, then 5ms fade in. Simple as that. If it doesn't work at 

your home, it's because you don't build it correctly.

If your ears are too delicate for this delay, it also depends how long 

is your sample. If you're playing a drum, building a minimum voices 

poly~ would never charge your cpu. And for drum, you don't even need 

anything else than a simple groove~ player AND the fade solution. If 

you're playing long waves, 24 poly voices will be ok. About commercial 

samplers, please note that, in the case of Kontakt, for example, there's 

ALWAYS a problem with voices, stealing & cpu charge. All my different 

versions of Kontakt begun to play shit after the 24 full voices were fed.

Forget about adsr~, it's not as good as it seems to be. Build your own 

enveloppe with line~ or, better, curve~. This last one will allow you to 

> please build me a patch! you clever people
>
I didn't give an answer to your last post because of this. Don't order 
people to 'show  a patch' as i usually do when asking for C code 
about delays ;-)

1 - you don't explain your problem _clearly
2 - you're way too much nervous & aggressive to give people the taste to 
help you.

Again, the 10ms solution is the best one in any case, mono or poly. 
Human usually detect a delay starting at 20ms... I use this solution 
since years, in *every* MSP patch i have, to avoid any kinds of clicks. 
5ms fade out, then 5ms fade in. Simple as that. If it doesn't work at 
your home, it's because you don't build it correctly.

If your ears are too delicate for this delay, it also depends how long 
is your sample. If you're playing a drum, building a minimum voices 
poly~ would never charge your cpu. And for drum, you don't even need 
anything else than a simple groove~ player AND the fade solution. If 
you're playing long waves, 24 poly voices will be ok. About commercial 
samplers, please note that, in the case of Kontakt, for example, there's 
ALWAYS a problem with voices, stealing & cpu charge. All my different 
versions of Kontakt begun to play shit after the 24 full voices were fed.

Forget about adsr~, it's not as good as it seems to be. Build your own 
enveloppe with line~ or, better, curve~. This last one will allow you to 
use a very handfull curve factor.

"please build me a patch" is not meant to be agressive.

its alot easier for me to understand a patch than a description of one. i share enough of my patches, such as they are. you can have the tracks i made with em for free too!

anyways, i agree adsr~ seems not as useful as building your own envelope... i couldnt get rid of clicks using adsr~ in a monophonic player

"please build me a patch" is not meant to be agressive.
its alot easier for me to understand a patch than a description of one. i share enough of my patches, such as they are. you can have the tracks i made with em for free too!

Ok. If you're ok in a mono world, here's what you need :

[ddg.mono] external by Darwin Grosse (you should have since your first 

This is an example patch built from yours :

> Quote: bin wrote on Thu, 14 December 2006 19:28

> ----------------------------------------------------

>> so basically youre suggesting introducing a latency into the sampler to allow time to fade out. is this what the exs24 or kontakt or whatever does? how come you cant hear a latency in those instruments?

> I assume these samplers are all polyphonic, which means they will always start a newly triggered sample on a new voice (just like poly~), fading out the previous sample independently from the new one in its own release phase. This is a feature that of course uses more cpu power than a monophonic approach. If there would be only one voice I assume you'd get clicks.

> Interactive audiovisual sampling soft- and hardware

[ddg.mono] external by Darwin Grosse (you should have since your first 
days, if not -> Maxobjects.com)

Mattijs Kneppers wrote:
> Quote: bin wrote on Thu, 14 December 2006 19:28
> ----------------------------------------------------
>
>> so basically youre suggesting introducing a latency into the sampler to allow time to fade out. is this what the exs24 or kontakt or whatever does? how come you cant hear a latency in those instruments?
>>
>>
>
> I assume these samplers are all polyphonic, which means they will always start a newly triggered sample on a new voice (just like poly~), fading out the previous sample independently from the new one in its own release phase. This is a feature that of course uses more cpu power than a monophonic approach. If there would be only one voice I assume you'd get clicks.
>
> Mattijs
> --
> SmadSteck - http://www.smadsteck.nl
> Interactive audiovisual sampling soft- and hardware
>
>
>


that fe.nocut~ patch is just what i needed...

thanks!
that fe.nocut~ patch is just what i needed...


use the pipe object to delay the midi note for 5ms or so, and have a bang go to the adsr~ to start from zero gain. Then the delayed midi note is processed.

how do samplers work?!