Satanic Smurf

I use BFD as a plug in with Pro Tools. I'm not sure what you are using but maybe this will help. In Pro Tools you create a stereo audio track and insert a plug in, BFD stereo. There are other options where you can create a seperate audio track for each kit piece but BFD stereo is the best way to start. Load a kit into BFD. Create a midi track and set the output of the midi track to BFD. Now when you click in the midi track with the pencil tool you should hear the kit pieces. If you click the icon with the piano keys in the BFD window it tells you which notes correspond to which kit piece. Set up a grid and begin the laborious process of programming drums. It doesn't have to be PT's either. BFD can be used as a VST plug in so I guess that includes a fair few programs like Cubase, etc. As far as other software to use with BFD, apart from the program you're using as a sequencer; you could run other plugins with BFD. They also have those refill packs for BFD like XFL and 8Bit. These aren't really other software though. I'm not sure what the sounds in these packs are like so I'd be interested to hear what other people think of them. i.e. is it worth the money? Sorry if this wasn't very helpful

Originally posted by UstadKhanAli If I understand this correctly....the "imaginary slider" is your way of saying "as if someone is pulling down the fader", right? I was refering to the simplification that Kiwiburger used earlier in this thread, where you imagine a compressor as an automation of a guy pulling down a fader when the singal gets too loud. Now we're getting onto envelopes it would seem that the subject is a tad more complicated then I had initially assumed. I've read a fair bit on compressors on the net but most of it is all the same oversimplified stuff. Do you know where I could get some detailed info on the subject? You know maths, diagrams, etc. Preferably web based (so I can slack off at work) and free ('cause they don't pay me much to slack off)

Thanks for clearing that up guys. Sorry Kiwi, I wasn't thinking about using a compressor for tracking. I've only got software compressors. Tracking with them would be sort of pointless becasue the signal has already gone through the A/D converters and like you said, you can't catch the transients anyway. There is normaly enough dynamic range in the digital world anyway. I forget how to work out the dB range for 64 bit but I remember that it was a lot. So my release time question is sorted. So lets move on to the ratio question I had. I asked this before but I'll type it again so you don't have to scroll. If the singal goes past the threshold (on it's way up) the imaginary slider goes down after the attack time has expired. For a given raito is the amount that the slider goes down fixed? OR if the volume were to increase further would our imaginary slider be moved down further? Thanks P.S. I love this board, I almost feel like I should be paying you guys for this info......but I'm not going to It's the thought that counts after all.

So what is the method that mastering engineers use to pump up the volume? I read a lot about music these days having no dynamic range. It's been squeezed. I always assumed that this was done with multiband compressors and limiters. Do they just bang on some saturation instead? Sorry if I keep pressing the same subject but I've thought one thing for a long time (fast attack compressors reduce transient peaks and give more head room). It's sorta hard to grasp then new way of thinking.

I once was blind but now I see Thanks for clearing that up. While I've never used compressors while tracking (I only have the plug in version that comes with pro tools - not much use while tracking). I've always been thinking about compressors the wrong way - as a means of reducing my transients. So I'm not sure if this needs a new thread or not but... What are the ways of reducing transients? I know from your earlier coments that saturation is one. Are there others?

Thanks for the info:) Now for more questions. I guess that this isn't strictly on the thread topic of release times but it's still compressor related: I was a little confused when you said that compressors didn't catch the initial transient. I thought that people used limiters (or compressors) when mastering in order to reduce the transients, thus giving more head room and allowing the volume to be increased before clipping. Is this a misconseption? Compressors usually have attack times Is there a way to use limiters to reduce transients without degrading the signal? Or do people mastering music accept this degredation in return for increased volume? In regards to saturation reducing transients: Could I try a cheap version and try running my tracks through amplitube to put a bit of drive on them to reduce transients? I don't really think that this would work but someone somewhere might have tried it and made something of it. Could you recomend a good free tape saturation plug-in (VST or RTAS)? I know I could find this myself but if you know of a good one offhand it would be better. OK, that's all I can come up with at the moment. I'm sure there are other questions floating around in my head but it's late in the day and work has turned my brain to mush:freak:. Sorry if these questions are incomprehensible.

Thanks Kiwiburger, I've read your post a few times over and I think the penny in beginning to drop. This is the second time you've helped me out so, cheers I appreciate it But while we are getting into the technicalities of compressors, I have more questions: Does the compressor reduce the signal above the threshold or the entire signal? Example: Ratio = 2:1 with the threshold set to 10dB. A signal enters the compressor that is 12dB. Is the entire signal, the 12dB reduced or is it just the 2dB above the threshold that is reduced? At what stage is the make-up gain applied? What I mean my this is: does the threshold take into account to the signal + the make-up gain? i.e (Gain x Signal + Signal)/compresion factor OR is it just the signal itself that is affected and the gain is added after? i.e Gain x Signal + Signal/compresion factor My formulas could be way off (and probably serve only to confuse). I'm assuming that the make up gain is just a scalar multiplier of the signal. Thanks.

The function of the release time on compressors continues to baffle me. In all the articles that I've read I get descriptions like: Deturmines how long the compressor continues to act after the signal has gone back under the threshold. I've also read that by increasing the release time it can increase the amount of sustain. So what does is meant by 'the compressor continues to act'? For example if I had a compressor with a ratio of 3:1 and a threshold of -10dB, an attack time of 10ms and a release time of 50ms. A signal comes into the compressor and goes above -10dB, 10ms later the compressor kicks in and starts reducing the signal as stipulated by the ratio. So a signal of -7dB becomes -9dB. Later the signal drops below the threshold of -10dB, what now? Is the signal still being reduced by the ratio of 3:1 for the next 50ms? If so, wouldn't this make the signal even more quiet How could that possibly improve sustain? I understand that by reducing the transient peaks we can boost the signal without introducing clipping distortion thus making the quieter parts of the signal louder. This is how a compressor adds sustain. What does this have to do with the release time? Also what happens if I recive anouther transient during the 50ms release time? I assume that the compressor just doesn't disengage