Hi, I've been using this script for a while to play wavefiles backwards and forwards, but realized that it just isn't doing what it should do. Basically, I have a message that specifies 0.0666537 for forwards and a message that specifies -0.0666537 for backwards (as an example). however, if i get sent a 2 bar or a 1 bar wavefile, these will result in, well, very low-tempo playback.

loadbang
|
symbol $1
|
read -resize $1 $1-L $1-R(
|
soundfiler
|
numberbox
cold inlet)
*~
|
tabread4~ $1-L
|
| tabread4~ $1-R
| |
outlet~
outlet~

second go:
loadbang
|
0.0666527 (
|
hslider
|
number
|
sig~
|
phasor~
hot inlet)
*~

sorry, not sure how to use the forum to explain such a (to me) complex script. anyway, i hope the .pd itself will make some sense.

basically, i have to slightly detune the loop to get forward / backward playback, and this is causing some issues. i'm not entirely sure how to read the samplesize filesize so that it can be used by the message-boxes for banging forward / backward playback..

if anyone has any ideas, please do let me know.

http://www.pdpatchrepo.info/hurleur/msx~.pd

soundfiler gives the size of the sound file in samples. to use it with phasor~ with no pitch shift you need to get the frequency for phasor~ by dividing the samplerate by the number of samples in the sound file, for instance with a samplerate of 44100 and a sound file with 22050 samples, the frequency of the phasor~ would be 2 because 44100/22050 = 2.
the value of 0.0666527 seems arbitrary, you would use that number if your sound file was about 15 seconds long and recorded at the same samplerate you were playing it back with

@sebfumaster said:

soundfiler gives the size of the sound file in samples. to use it with phasor~ with no pitch shift you need to get the frequency for phasor~ by dividing the samplerate by the number of samples in the sound file, for instance with a samplerate of 44100 and a sound file with 22050 samples, the frequency of the phasor~ would be 2 because 44100/22050 = 2.
the value of 0.0666527 seems arbitrary, you would use that number if your sound file was about 15 seconds long and recorded at the same samplerate you were playing it back with

hi, so basically i would be using the number output of soundfiler, dividing 44100 with it.. and then inverting that number and banging that to reverse playback, and using the non-inverted number to have forward playback? did i understand it correctly?

yes, not inversion in the mathematical sense but rather a sign change (multiplication by -1)