[LAU] What would you recommend for fine-tune equalization/calibration?

I am in process of building a linux DAW system. I'm planning on using a bunch of very cheap speakers for playback of the middle range (each with its own channel), along with a stereo pair for fairly decent low range speaker for the low freq <150 hz (15" with a 5 lb magnet) and fairly decent high end tweeters for >4khz. The low freq and high freq channels will come off of an electronic crossover and can be adjusted independently from the middle range speakers. I'd like to get some method of injecting pink noise and auto-calibrating the midrange speakers to pre-compensate for the 'smile' (or frown) shaped frequency response curve of the cheap-o speakers. The speaker/amp setup will remain in the same place for the duration, so the calibration will only need to be done once (or until the room, the speakers, the amplifiers, etc change) What would you recommend for: 1. Applying a pink noise to everything 2. Auto-calibrating (pre-warping) with fairly narrow freq band filters (at least 3 filters ~+-10 db per octave) This doesn't have to be done real time. It is anticipated that I'll want to run 'wav' or similar files thru the filter prior to doing some additional 'black magic' with delays on various sound sources. Even though the stereo crossover and existing equalizer has some capability for doing this, but it may be easier to do everything with the same method and handle all of them at once with the same method. I'm looking at various DSSI, LADSPA, PD, etc effects to do this, but I'd like to work with what someone else has already had some success with. There is an extra bonus if I can use the same software suite to add in a variable amount of delay per source, since this will be the next step. There is a high likelihood I'll go with a method that is recommended by someone and has shown good/positive experience. I'm currently thinking I will need to script the part where a variable delay is added, and simply add in a number of '0' samples to the front end and back end of a source (to keep everthing the same length). Depending on how far the sound source is from each speaker, I'll put more 0's on the front of the sample list for sounds that are far from the speaker and less 0's on the front of the sample list for sounds that are close to the speaker, with the difference going to the back end of the list. What I will end up with is a bunch of files that are almost the same thing, except they will have a variety of delays in the start time. I'll then add up all the sources with Ardour and have a separate mix per speaker. TIA, Mike Mazarick