crispypancakes

i have built an FIR filter and it seems to be running well, it makes the sound much larger than when it's off, is this supposed to be happening?

theres a lot of literature on how to build one, hence i have built one, however not a lot of explanation on what it actually is meant to be doing.

can anyone explain this in lay terms so i know whether mine is working ok?

this is an experiment rather than an essential tool.

i have built an FIR filter and it seems to be running well, it makes the sound much larger than when it's off, is this supposed to be happening? 

theres a lot of literature on how to build one, hence i have built one, however not a lot of explanation on what it actually is meant to be doing. 

this is an experiment rather than an essential tool.


fir-filter

Your way does not really work because tapout~ has a minimum delay time of one signal vector.

or buffir! there is a good example in the examples/filter/buffir folder

so without using a single object and by modifying the patch i put here, how would I change this to work correctly? can it be done?

as a didactic exercise, this is useful although there are good examples in the, erm, examples folder as oli suggests. The FIR filter effect is heard when the delayed samples are summed with the current samples. Each of the 4 tapout~'s are scaled by division by 4; then divide the source and the delayed signals each by 2 before summing - this should work. I made some simple changes to your patch, with notes (including Chris' reference to the signal vector size), to allow you to hear the filtering effect of the delay lines.

Ultimately, the effect of this FIR is largely dependent upon the nature of the soure material.

Hi
as a didactic exercise, this is useful although there are good examples in the, erm, examples folder as oli suggests. The FIR filter effect is heard when the delayed samples are summed with the current samples. Each of the 4 tapout~'s are scaled by division by 4; then divide the source and the delayed signals each by 2 before summing - this should work. I made some simple changes to your patch, with notes (including Chris' reference to the signal vector size), to allow you to hear the filtering effect of the delay lines.

what vector size should i be aiming for? is there a measurement to ensure this is the correct size?

the rest of the patch i understand, and thanks for taking the time to explain it.

so where inlet 1 is the synth signal from the oscillator and inlet 2 controls the gate switch, is this more like the idea i should be going for? 

i found the following part of the patch from one of max tutorials, where i can put external filter co-efficients to generate FIR filters, but i can not use it to filter my audio signals. any idea how to solve this problem and how use the co-efficients from this dictionary object to filter signals?