Hi it's my first post and first big project so sorry if any of this is basic. I am currently building a step sequence sampler with a multi effects rack and mixer but having problems with my Aux sends. I am using throw~ and catch~ to create a basic summing bus for each Aux. They are 100% wet buses so no aux returns - the output goes directly to dac~. When they are engaged and running through the effect, it creates a phase/comb filtering problem with the signal - I'm guessing because of a processing delay? A basic phase inversion switch didn't solve the problem thought it was worth a try... Anybody know how to create a delay compensation set up or any other ideas as to how to resolve this?
Delay/phase problems with Aux sends and effects
You are probably correct about the delay, but you must be sending the same signal twice (one of those times with more delay) in order to create such artifacts. The send~ and receive~ objects introduce one block of delay, and I think that throw~ and catch~ do as well, so if you are throwing and catching more in one path than in another then you will have delay differences for sure.
Thanx for the response, is it 1 block per throw and catch or will a subpatch with multiple throws/catches, sends/receives be delayed by 1 block to process all of them? Have tried matching the dry signal using a delay object with no effect
http://puredata.info/docs/manuals/pd/x2.htm.... you will find the answers in paragraph 2.4.5 ! and it should be easier to spot the problems in your patch. When a signal is "sent" it doesn't "arrive" until 1 block later. Each time you pick it up and throw it again you lose another block. Of course normally you don't hear any artefacts because you don't mix the same signal (without its delay) back in; unless you want that effect! You must have the same source arriving twice at the dac~ for the phasing to be audible......unless you have a live source??
If you have a live source (a microphone) and your patch is using that source, processing it and feeding it back to some speakers then you need to reduce the delay for any signal to the lowest possible or you will hear the delay. 2ms is as low as most modern computers can manage with "asio" (that's the same as a Yamaha Digital desk). At 2 ms a violinist with an earphone will not hear the delay between the instrument and the earpiece. At 5 ms they probably will. Try "2" for delay in the media setup and 64 for the block size...... see if you get any dropouts. The smaller the "delay" the smaller the "block" size, and so the smaller the delay for every send/receive.
Later, when you have some free time you should read all of the link above....... it gives a useful and fascinating insight into what is going on "behind the scenes".