i am pretty new to pd and am making a looping sampler that takes in audio from adc~ and automatically loops it... however, I can only seem to loop the first second of my sample regardless of the length of the original sample. I know I must be overlooking something, so any help would be great. Thanks!
- 
				
				
				
				Read array 
 
- 
				
				
				
				Check out example B.08.sampler.loop (in Help>Browser>Audio Examples) and the other examples provided. 
- 
				
				
				
				assuming you have a samplerate of 44100hz, then try lookign through your patch for objects, tables, arrays or messages containing the number 44100. this is the amount of samples in 1 second of audio. you will need to make this number bigger if you want more than 1 second. 
- 
				
				
				
				Along the same path of reading array's. 
 Heres my question- Is it possible for say once you have imported a wav file into an array you get the drawing of the wave form etc....but is there a way for pd to automatically read the length of the imported file and produce the length eg time in seconds of it in a number box ?
 Hope that makes sence.
- 
				
				
				
				the outlet of [soundfiler] will give you the length of the sound in samples. just divide this by 44100 to get the time in seconds. there is also an [arraysize] external that will tell you the size of the array in sampels when banged. 
- 
				
				
				
				Since were on the topic, is it possible to find out the samplerate of the loaded file, because i still work with a mix of 44.1 and 48 files. The division by 44.1 is actually the division by the current pd samplerate, right? |] [] |.| ][|-| -- http://soundcloud.com/domxh 
- 
				
				
				
				there is an external called [soundilfe_info] in the iem library, but it doesn't seem to like my .aif files. works ok with .wavs though it seems. 
- 
				
				
				
				Thanks hardoff, works fine here with .wav files. |] [] |.| ][|-| -- http://soundcloud.com/domxh 
- 
				
				
				
				cabin2j, 
 did you ever resolve this issue? I am having the same problem, and can't seem to get out of it...playing around with example b.10 and can't get the phasor out of 1 second cycle...
 dbedit: example b.09 db 
- 
				
				
				
				the array in the example is only 441003 samples, so it's cut off at one second. If youse guys are looking to do "phrase sampling" , like in Ableton, try using [delwrite~] and [delread~] instead. 
 
					