ATTN: Buzz users

it's a really odd error. it's complaining windows\system32\dsplib.dll is missing

which it is. might be this weird stripped down build of xp i'm using

some feel like pm'ing me that file?

nevermind, i figured it out, for some reason i have to copy it from my buzz dir to system32 :undecided:

so the beta decided to inexplicably NOT crash for the first time ever tonight

*cums* its got loads of stuff i wanted, and the crash and missing features are fucking GODLIKE

it seems really stable so far (once it launched lol) and is it me or is the audio quality a smidgin better?

found this kickass tutorial on a random forum

really is the sort of thing should come with the beast:

Okay, time for some more Buzz talk. Settle down and listen to an old man whine.

First of all, let's go through the note tempo thingy. My patterns are always 64 ticks long, a remnant from my past Scream Tracker days. What this means is that basically the songs are 4/4 (except for my random forays into the worlds of 3/4 and 7/8, in which case my patterns are 48 and 56 ticks long, respectively). This means that with a standard about 130 bpm, if I want a 4/4 bass drum, I play a bassdrum hit every four ticks. However, I usually set the song's speed to 7 or 8, which means I would program the bass drum to hit every 8 ticks instead of four. This method gives me more ticks between a time sequence, and enables me to make faster hihat riffs or nice, long and tight snare slides. I just like having extra ticks to play with.

Oh yeah, hitting CTRL+backspace in pattern view enables you to change the amount of ticks per pattern. The number is pattern specific, so not every pattern must be the same. This enables the user to build songs that change from 4/4 to 3/4 and back if needed. Also, to get a good view of what's going on in the sequencer view, set the displayed ticks there to correspond with the number of ticks that are in the majority of your patterns. That way you can see one pattern per row in the sequencer, and control your song more easily. No matter how many ticks you want per pattern, always make sure that the amount is consistent with the tempo. For example, both 64 and 16 are divisible by four and thus usable in 4/4 or 8/4 or whatever mode, while 48 is divisable by three, and thus okay for 3/4.

With that out of the way, let's go into the mixdown:

Assuming you've already got a few machines connected to the Master, go check out the Master properties by right-clicking on it and selecting properties. Now you'll see the volumes at which the master tape receives the instruments. NEVER, EVER, when beginning a song and only having a few instruments, keep all the volumes at 100%. Move them down to about 75 or 80%. This'll give you some headroom on the mix, which is very important when you add more instruments.

Always monitor the master volume level (as seen in the upper left corner) and don't let it max out if you've only got a couple of instruments playing. If you leave no volume headroom at all, you'll have to resort to heavy compression when mastering the track, which robs the music of all its power to truly kick ***.

Here's a couple of tips on how to EQ instruments properly:

1) Using a graphic EQ to boost a certain frequency is a bad, bad idea. Always use graphic EQ to cut undesired frequencies 400Hz to 1kHz is usually where all mix clutter (useless extra sounds) end up in, so cutting a bit there is likely to make your song sound clearer.

2) For boosting a certain frequency range, the parametric EQ is your mate. Find tutorials on the net to understand its mechanics better, but in short it gives you much more accuracy in dealing with both wanted and unwanted frequencies.

3) Use EQ to boost a frequency only when it's absolutely necessary. If your bassdrum or bass riff is a trebleish whine from the start, pumping up the low end isn't going to make it any better. The best thing to do is to switch to a sound that's better right from the start. Using EQs can make a good sound godlike, but can't make a crappy sound good. Instead of boosting, cut the frequencies you don't want to hear and raise the volume of the instrument instead. I usually only boost certain low level bass frequencies to give my bass riffs extra richness.

4) A good idea is to open one extra graphic EQ and connect all instrument/effect chains to it instead of the Master. Then connect the graphic EQ to the master. This gives you an EQ that effects the whole track, and can help you clear mix clutter. If some frequency range just sounds plain awful and you want to cut it from the song entirely, attach a parametric EQ between the graphic one and the Master, and find the offending frequency range. Then cut it down.

5) Kill useless frequencies. For example, if your hihat samples have a funny peak at bass frequencies that you don't really need, cut it out of the mix with an EQ. However, remember that harmonics always go upwards, so never kill high frequencies from bass sounds, unless you're absolutely sure that the instrument will sound better without them.

Now, what the heck is a compressor anyway?

A compressor is in short a tool that controls the volume of a single instrument. If you want to use a compressor, always place it between the last effect in a generator/effect chain and the Master. Take a look at the picture of the machine view I posted on the second page of this thread. You'll see a lot of machines titled jComp as the last effects in a chain. They are all compressors set to limit the input from the machines before them.

How to set one up and what exactly does it do:

1) Set the compressor's threshold to -0dB (100%)

2) Set the compression ratio to 1:1

3) Set the attack time to as low as possible

4) Set the release time to about 15-30ms

Threshold is the point where the compressor starts working. If the sound passess the set threshold point, the compressor starts cutting it down. The compression ratio means how much the sound will be cut when it reaches the threshold. 1:1 would mean it won't be cut at all. 1:2 would mean it's volume past the threshold will be halved, and so on. Attack time means how fast after the sound has passed the threshold will the compression start taking place, and release naturally is the lenght of compression after the attack. By setting the release time to a very short interval, the compressor will only cut at the points where the sound is louder than the threshold, and leave the quieter parts of it alone.

If you want to record songs at loud volume levels (is there a point in recording quiet?), learn the art of the compressor and its sibling, the sidechain compressor.

Okay, now what exactly is the point of all this talk?

I've heard a lot of demos and non-professional songs in my time. Most of the time the artist has mixed and mastered the songs him/herself, and the result can be heard easily. Muddy mixdown (bad sound) is something that happens to everyone, especially if your home speakers are teh suck. I don't have superfine monitor speakers, so I always listen to the songs in three ways before deciding if a mix sounds good. These three ways are stereo speakers, headphones through the stereo amplifier, and headphones straight from the sound card without anything in between. The mix should sound good or at least ok on all three.

However, nobody can mix at home like they can do at professional studios, so do please compensate by recording loud and clear. When a digital sound reaches the machine's peak level, it will start chopping up or distorting. Most of the time this is a very bad thing and should be avoided at all costs, unless you are Thomas Garrison of Control, who uses the chopped up effect to add even more violence into his sound.

Most people instinctively lower the master sound level if it peaks out. THIS IS NOT THE WAY TO GO. The way to go is to find the offending instruments which produce peaks in the mix, and compressing them a bit. If that doesn't help, lower all individual instruments' volumes in the Master, but don't you dare touch that master volume level. For the record, I do not record loud enough yet, but plan to hone my skills until I do.

Overcompression is always a bad idea, unless you plan to produce radio-friendly, sucktastic pop music. Ever wondered why even the most aggressive drums like the snare sound so bland on pop songs where there is a lot of distortion (ie. heavy guitars) present? Compression, baby. The producer has decided to compensate for the heavy distortion, which is bound to cause a lot of peaks in the mix, by compressing all peaks out, leaving a lifeless husk of a sound that doesn't offend anyone, but nor does it have any soul or energy in it. The point of distortion is to make sounds aggressive, so you should always seek a way to make it louder and still keeping the mix under control, instead of just copping out and compressing it until it loses all its dynamics and pow

:|

:)

good tutorial

ps i'm still stuck with the old buzz version because there still isn't a midi clock slave feature in the new one :|

anyway buzz is still great, i keep discovering stuff regularly.

:)

LOTS of new updates! new effects, an installer too (in the 'setup' folder)

http://jeskola.net/buzz/beta/files/

I know, the update rate on this 'un is bloody grand. The loading time has kinda trebled for me over last few releases for some reason...

My current buzz looks like this:

The changes are mainly functional changes, visual tweaks and bug fixes (lots of bug fixes) than anything else. Sound wise the only difference is the optional oversampling and 32bit sound recorder - Here's a link to the major changes - http://www.buzzmachines.com/viewreview.php?id=1192

God if Buzz2 finally comes out bug free and rock solid I might have to buy a PC laptop

  ten fingers ten toes said:

The point is simply that, as I originally stated, that Renoise is more accurate when rendering to wav. I said 4 posts back that sure you can get great sounding results out of Buzz, but since my original statement of Renoise's superior audio engine was challenged, I supported my statement. You basically can't render to wav more acurately than Arguru's sync in Renoise.

"accurate" means nothing, as far as timing goes, you may be right, although I assume they are fairly equivalent. As far as resampling, that's entirely generator dependent. Buzz doesn't have a sample tracker built in, the trackers(samplers) are made by users. Resample accuracy shouldn't even be compared when talking about the buzz environment itself. Rendering is done like any other DAW that works in 32 bit floating point.

EDIT: also RIP arguru. :unhappy:

Yeah I was trying to explain him that a page or so back (until the selfish bastard decided to get himself banned :laughing: ). He didn't seem to grasp that Matilde Tracker was not the only thing that played back samples in Buzz !

nice

http://www.buzzchurch.com/viewtopic.php?t=3404

i'm still using original buzz. don't experiment much with machines i don't know so dont crash much.

doubt i'll update till i get a new laptop.

  qnio said:

nice

 

http://www.buzzchurch.com/viewtopic.php?t=3404

Woah, that looks ace. oskari is making some cracking progress here.

how are people finding MIDI support in the new version?

it doesn't like my USB controller at all - it causes it to crash, but in a recoverable manner