sample based phasor problem (synchronization sollution)

    Feb 18 2006 | 9:55 pm
    i am trying to create patch which will combine live sampling and song playing (as kind of playback) at the same time and i am solving synchronization problem.
    my idea is - groove~ is playing "the song", and is synced to the wave~ which plays one channel wav which generates master phasor, because content of this track are phasor waves synced to the bars of the song. eg. phasor goes from 0 to 1 every 8 bars. the advantage of the usage of the sample based phasor is that i am using iregural bar structure so it seems easier to just "manually describe" structure of the song.
    the problem is that the max doesn

    • Feb 19 2006 | 6:55 am
      tomas dvorak wrote: > my idea is - groove~ is playing "the song", and is synced to the > wave~ which plays one channel wav which generates master phasor, > > the sounds synchronised in this way seems to be modulated kind of > lo-fized. also analysation objects like change~ seem to get crazy > from it. i have feelings it could be problem with the dataformat.
      How long does the phasor need to get from 0 to 1? It sounds as if your construction is getting into 32bit floating point resolution limits. If you want to span more than 6 minutes to navigate with a phasor~/wave~ combination you need to use the hr. objects from Joshua Kit Clayton. (hr.phasor~/hr.wave~)
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    • Feb 19 2006 | 11:57 am
      stefan, thank you very much for your suggestion. the sample was really longer (about 6.30), i tried to shorthen the file on around 2 minutes, unfortunatelly the problem persisted.
      i would be happy for any other idea or help .o)
    • Feb 19 2006 | 12:32 pm
      Hi tomas,
      can you send an example?
    • Feb 21 2006 | 5:48 pm
      hallo, thank your reply, i made two schemes of what i am trying to do, however any of them doesn
    • Feb 22 2006 | 1:21 pm
      hallo, i am still solving the problem with the sample phasor based synchronization.
      i made small and clear example patch, so the problem i am solving will be easier to see. it is simple player of the sample based phasor (which is also attached). if you play it, change~ detects the direction change hunderds of times per second, also syncing wave~ or play~ to such phasor causes signal degradation.
      seems to me like there would be some problem in the internal msp bit depth and sample rate and the format of the wav. i tried many different wav formats (22, 44, 96 khz and 16, 32 bit combinations). unfortunatelly any of them haven
    • Feb 22 2006 | 4:16 pm
      hi Tomas,
      the soundfile in your example is int16, i think this is where the degradation comes from, try recording as float32 (= Max internal precision). See patch for taking a close look at signals with capture~. Dunno about high-res.
      That being said: You never mentioned what the phasor~ does or how it is generated. Might be simpler to generate on the fly than using a prerecorded one, no? Something like triggering groove~ for the soundfile and zigzag~ for the ramp(s) ...
      g, g.
    • Feb 24 2006 | 1:56 pm
      hallo grg,
      thank you very very much for your post. it was very helpfull. finally i found what is the problem, it was of course my stupidity .o) i was converting big-depth in the sound editor trying which one is the right one. even if it was 32 bit float due to conversion the file was bad despite format was ok. the problem was i wasn