I have a .wav file that I have opened in CHOPs, and I am trying to get numbers from the levels of the frequencies. I would like to have it divided into about 28 bands, with a variable representing each. As the track plays, the value of each variable would change according to the frequencies level at that point in time. This is my first time in CHOPs, and I have only the help screens to learn from. The most promising way that I have found thus far would be to use a Constant CHOP to create as many band variables as I choose, and within each variable write an expression that would single out a certain freq. range and derive its value from it. My only problem here is that i dont have any idea of the syntax to use to look at different freq. of a wave.

Use a Spectrum CHOP to compute the frequency spectrum, and then a Resample CHOP to resample the data (select method = “New Rate, Same Time Range”). I had to manually adjust the sample rate until I got 28 samples for the one time range; 120 is a good starting point, and work upwards (for 44.1kHz and 24 fps, 265sps worked).

There are also local variables with the band-eq and parametric-eq chop. Any frequencies spread across different channels should be easily separated that way as well.

Use a fan chop to make lots of channels, math to set to one, then math to multiply that by the .wav file.

After that send those .wav files into the EQ etc and extract the frequencies according to HZ. You might also want to check out “fletcher munson curves”

http://hyperphysics.phy-astr.gsu.edu/hbase/sound/eqloud.html [hyperphysics.phy-astr.gsu.edu]

most of our hearing happens between 200-2000 hz, with a spike at 4000. If the frequencies are separated equally than there will be a lot of channels that really won't correspond with what people think they are hearing

andrew i ended up using a modified version of your approach. whenever i used a fan CHOP is started to crash my machine or bog it down. so i used constant to create all the channels and set them to one, before i multiplied them by the .wav. im going to try with default 14 channels that an eq. gives, and if the visuals dont correspond, i will change up the freq. thank you both

memory is always a problem when dealing with Audio and chops, the only way to make it better right now (I think). Is to use the “unload” button. This will disguard the memory once the chop is done processing it.

There is also the “time slice” but this only works for certain things. also if you don't need for fan to output at 44100, set it to something normal like 41 or 24 depending on your setup.