I've got kind of a strange problem, when I open an audio file into readsf~ it always plays a little bit pitched down and slowed down, as if it's playing at the wrong sample rate. I've checked pure datas test tone and it played the correct sine pitch.I've checked the audio file in a number of media players and pure data is the only one that plays out of tune.
I'm running pd 0.43.4-extended on windows 7 using the default MMIO at 44100 with a 64 block size.
my code is simply:
[open SONG1,wav, 2(
|
[readsf~ 2]
|
[output~]

any ideas? :3

first idea was sampling rate.
second idea was that pd-extended has a bug.

Try that in vanilla and see what happens?

My guess would still be sample rate. If your original was at 48k, most players that I know of would automatically adjust and play at the appropriate sample rate. But PD would still be playing according to its settings, ie, 44.1 in your case. So a little slow and out of tune makes sense to me.

You could check the sample rate, or just save a copy at 44.1 in Audacity or some such to see if that is the case.

Had a similar issue with slowed down video which uses the the sample rate of the stripped and now separate audio file which is to control the frame rate of playback of the video ......this may sound confusing but have a look at how the audio playback is set up in this patch and it may help.....as usual there is a little math involved but have a look .....1472220530700-av-sync.pd

@lilsquiddy If you are lucky enough to have stubbornly stuck with PDextended, or you have "dekened" your vanilla you could do this..... although it only helps for one track at a time.......
dunno_what_rate.pd
@WilliJ.... did my av-sync solve your problem? even with the x-fade?
David.

found a simpl solution to the problem: converting my track into 44100k, 16 bit. My wave file was actually in 44100k, 32 bit. I'm guessing what happened was pd "stretched" the file to fit 44100k 16, causing the distortion...
not really sure why i was compiling audio at such a weird ratio in the first place lol.

@lilsquiddy Yes, 32-bit floating point is an option in most audio recording software........ and default in some.
I am not absolutely certain, but looking back at your first post I see that you sent the message [open file 2( to [readsf~] and it could be that it took the 2 to mean 2 bytes (=16bit)...
If you remove the "2" from the message it might read the header correctly and then play correctly........
I would be interested to know!
David.