/ Audio/ MC-505 Tips/ Miscellaneous Tips |
THOUGHTS ABOUT THE TURNTABLE EMULATION IN SYSEX [Nina]
i was thinking about the turntable emulation and was wondering how this is done at the 505.
so i started cakewalk and recordes the turntable d-beam
the result was a sysex:
F0 41 10 3A 12 70 03 37 3C 1A F7
important is 37 3C =14140 =141.40
and for all parts the keyshift (cc 85)
this must allow (so was i thinking by myselft) to speed up the pitch and the speed and not only downwards...
SONG CREATION [Oliver]
I want to add some basics for song creation and mastering :
First of all its important that you know , that every instrument in a song should have its own room (not meant as an own different "reverb" which is often done by profis to make the instruments work together) . I mean an frequency part which every intstrument should have for itself . This begins in song programing where you play some instruments in deeper regions and some in higher but mostly in different ones . Then you have to choose the instruments: And i think most people underestimate the importance to adjust the sounds specific to a song . This means a pad can sound wonderful alone when it has deep frequencies but together with a bass they cover each other and sound not clear in the ear . So begin to adjust sounds with your synth filters as good as it goes because a filter will always sound better then your mixdesks equalizers . A patch mustnt sound that good alone , it has to work together with the others . Thats the point . To find the patches that work well together is one of the most important and most difficult things . Dont hesitate to change existing sounds for the songs needs.
The trick to make a clear mix is that every part has its own frequency and/or overtone range . This sounds easier as it is . Many sound engineers work with frequency analysis to set the frequencies properly but i think thats too much of work , trust your ears instead . Another trick is to give the instruments different reflections means reverbs , but this means too that you need external effects . Mostly it is better to not overuse your mixdesks EQs too much they can decrease the mono compatibility due lack of quality . I know pros which never use their mixingdesks EQs . Another thing too look at is to set the position of your instrument in stereo position . Set some instruments a bit to the right and others a bit to the left . For the drums you can use as better extreme positions as higher the sound frequencies are , this can be for example the drum in center , closed hihat full left and clap full right and open hihat a bit right and cymbals a bit left . Important is that you get the real wiiiiiiiide stereo sounds only when you set them properly to the channels in your mixing desk . This would be either a stereo channel or two mono channels with the pan of one set full to the right and the other full to the left (never both in center). Ok thats it for the moment , i hope you had fun with my english ;)
EQ, PAN, EFFECTS - WHEN TO APPLY [drK]
> 1. When recording your stuff
> initially is it best to record it mostly flat and
> center channel, then tweak the eq and pan when mixing
> down? Or get everything set EQ and Pan wise as you go
> along. I'm thinking the latter might be the better
> approach?
Two different schools of thought here. Some like to get it down right at first take, others literally fix it all in the mix. I tend more towards the second because EQ is pretty hard to undo once it is there and the use of EQ is very situational depending on the context of the sound and the intended use in the composition - all things that tend to be late binding in my own work. So I generally work with the EQ and effects after the tracks are down.
I do believe that you should get the ram track right though in that it sould be a good recording. For acoustic or vocal recording it is probably very important to get it right as it will warp your sense of the mix later if you don't. Also for electronic/synthesized sounds sometimes the effects and EQ are part of the patch so that should be respected. So my rule of thumb is to get the sound recorded as I intend it and view the adding of EQ and effects later as part of the mixing and balancing process. I know that may sound confusing (like when is it part of the sound and when is it a mix-fix?) but you develop a sense of it. Also you learn how much to apply at each stage and what. For example while I rarely boost the bass of a track I will commonly boost the fundemental frequency of a kick drum. Adding delay effects to a lead may be just the sound I need but I hold back on applying reverb to tracks.
NORMALIZATION & MASTERING [drK]
Normalizing is deceptively useless in most mixing situations as it will only boost the gain of the sample so that the highest peak just fits within the dynamic range (i.e. 16 bits in a 16 bit system). This really doesn't do much practically speaking because as mentioned high level transients, even very short ones only ocurring once, will establish the maximum value and nothing can go above that. Imagine what an inaudible pop at the very beginning would do - it would basically disable the whole effect of the normalizing. Secondly, normalizing does not improve the signal to noise of the recording.
Compressing the signal before recording can help as mentioned before. First, be very leery of using the bassboost feature on the MC505 because it causes an inordinate amount of bass gain to be thrown at the overall recording - like heavy bass EQ. Leave it off when recording.
I would suggest identifying the parts of the composition that present challenges to the dynamic range. Drums are useually the worst for this, especially the kick. Record those either as separate tracks if you are using a multitracker or bring them out of the MC505 using the four separate outs. However you do this what you want to do is add compression to these types of signals. Sart with a low setting of the ratio and work up to taste. if you record using software and do so with the track separate you could use a software compressor or plugin to treat it during or after - as long as it is done before the track is remixed with the rest.
When doing the final mixdown I find adding two things generally help the overall feel of the mix. One is to add ever so slight compression to the whole mix, the second is to add a bit of reverb using a good quality reverb. Not much of either, just below the perceivable level. It tends to smooth things out.
WHY YOU WANT TO ALWAYS RECORD AS CLOSE TO 0 DB AS POSSIBLE [J]
> I'm not a scientist so don't ask me exactly why (something 'bout a wider
> ranger of soundspectrum for your ears I believe) but it's the truth and
> nothing but the truth...:-)
[Talking about why you want to record as close to 0 dB as possible]
Well sound (in digital media) is made of a bunch of discrete SAMPLES, each taken at a moment in time. For sound this is a measure of the air pressure flying through the air (at the microphone, say) at the moment the sample is taken. This is plotted in the Y (vertical) axis. The X (horizontal) axis is time (from left to right, as per the norm). Now, this is exactly what you are looking at when you look at a waveform in <insert fav wave editing proggy here>. From left to right you can "read" the air pressure that would be flying through the air were you to hook up speakers and play back the waveform.
Now, an analog wave is a continuous line. A digital wave is like a bunch of DOTS on this line, because digital cannot continuously measure the analog wave. Instead what it does is take "snapshots" of the air pressure many times per second (44-thousand for sounds recorded @ 44 kHz) and assign to each of these "sound points" or "snapshots" a value representing the air pressure. Sort of like when you watch a digital watch, what it is really doing (well, bear with the analogy) is taking snapshots of what an analog one would be reading out, once per second. You can't see 15 seconds and a half on a digital watch because it doesn't "snapshot" in between seconds. Instead it just reads out the time every second. (I'm merging the two axes so it's a poor analogy but hey ...).
Okay on to the meat. So you have an analog wave of <insert fav synth sound> flying through the air. Me, I prefer the 505 "Waaa!" sound so we'll use this. So Waaaa! is flying through the air, and your microphone picks up the analog wave. It converts the analog wave into an analog voltage in its wire, and conducts this to your computer. The ANALOG/DIGITAL converter on <insert fav soundcard here> then looks at this waveform 44-thousand times per second and assigns a value to the air pressure (remember, now converted to voltage) in the wire at that moment.
Now, when you record values you can say it is 50 <units> or you can say it is 50.00001 <units>. This is what is called RESOLUTION, and is what you are talking about when you hear 8-bit versus 16-bit versus 24-bit. (You'll have to know a little bit of binary here so let's just go through it while we're sitting here and comfortable). Each bit in a digital system represents the presence or absence of electricity in a circuit (basically). It therefore can be off or on. There can be, or there can not be. Very Zen. Lala. So people realized off and on can only take you so far, so they organize these off/on pairs into BYTES of 8 BITS. When you merge bits in this way, you can then assign values to each bit, and represent any number you want with these bits. It's a base-2 number system. Lala, whoa I lost some of you.
[ASIDE: BASE-10 SYSTEM]
What does "125" really mean? Well basically our number system is base-10 which means each column represents a ten-fold increase over the previous column. Soooo, 125 is made of three columns. these columns are:
10^2 (100) | 10^1 (10) | 10^0 (1)
and the actual NUMBERS (that is: 1, 2, and 5) are just counting HOW MANY units of each column there are
Now it will seem stupid to do this because we think base-10. Going back to base-2
A binary BYTE: 0 0 1 0 0 1 1 1 Represents this:2^x 7 6 5 4 3 2 1 0
So we'll now break that number down:
[ASIDE OVER]
So note that the maximum you can represent with 8-bits is 256, and you get this by 11111111 = 128+64+32+16+8+4+2+1=256
Now, you're happily recording the voltages 44-thousand times per second at 8-bit (for simplicity's sake, and hell - you ARE only recording the damned Waaa! sound anyway! So who needs quality?). So you're going along, and you're happy. And the sound is getting louder and louder.
00111010
01010110
01111010
10010010
11010111
11111111 - whoa!
You're at max! Basically if the sound gets any louder (the air pressure any more pressure-ific or the voltage any more voltage-astic), you basically cannot record it! How can you get higher than 11111111? There's no more room, and you can't just ADD bits to your bitstream the way we can on paper. Basically, you've maxed out your recording system and that's that. This is digital clipping, and if you imagine a wave that is getting louder when you hit this phenomenon you'll know why it is clipping. (monospaced fonts everybody, we're going to ASCII ...)
/--\
/ \
<clip level>. . . ________. . . . . . . . . . . . . . . . . . . . . .
/ \
/ \
--/ \etc. on with the Waaaa!
/
/
____/-----/
so instead of getting the nice rounded peak you should (top line), you get the flattened peak (bottom line). Along with distortion, since that's not what the wave is supposed to be shaped like.
Now, getting to the actual question/point: you want to record very CLOSE to the max you can, because otherwise you are wasting bits. Let me illustrate. First we shall record a quiet Waaaa!, then we shall record a loud Waaaa! (but not clipped). Let's check the bitstream. (Numbers are entirely made up and for illustrative purposes. But don't go hangin' em on yer wall - they ain't THAT illustrative)
QUIET - LOUD
00000001 - 00010000 (sound increasing)
00000010 - 00010100
00000100 - 00011100
00001010 - 00101101
00010101 - 01110110
00011110 - 10110010 (peak about here)
etc. etc.
Now check what you have used. On the right you have a perfect recording. You have used all your bits to get maximum resolution, but on the other hand no digital clipping. On the left, however, you have only used 5-bits. That's 2^5 possible values for the voltage measurements. That means that the right hand one can record 256 different values (let's look at it that the computer can note a value between 0-255 for the voltage at every point) but the left hand one can only record 32 different values (the computer can note a value between 0-31 for the voltage at every point).
Obviously when you play connect-the-dots (which is what your DIGITAL/ANALOG converter does to convert the waveform in your <insert fav wave editor here> back to sound from your speakers, the wave which was originally recorded too quiet will only have used 31 steps in recording the wave. Your two problems will be:
So avoid these two problems, and record your waves as close to 0dB (which in a digital system represents full bits, or 11111111 in our example) as possible. LAla. Tada. Gotta go do w3rk now.
IF YOUR 505 CONTINUALLY LOADS THE WRONG PATCHES [rEalm]
[Original message:] So, a few weeks later, when I was able to return to me studio, I discovered that I could reassign the correct patches to the patterns by being in patch mode and pushing the PRESET, or USER buttons until the patch I had originally used was restored. In other words, I may have been using ptch:U:A001, but the machine substitutes ptch P:A001, or P:B001, or P:C001!
[Reply:] Ah Ha!!!! Been there before. You need to set your MIDI Program and Bank Select Rx settings in the systems settings to ON. If you set it to off (which seems logical), the pattern will keep using the same patterns, or the wrong banks for all of your patterns. Want a quick fix after you change those settings? Power on your 505 while holding SHIFT, then scroll until it says PATCH RESET. Then press Enter and wait about 45 seconds.
WARNING!!!!!
This will erase any patches you have stored in the USER memory!!!! Make sure you've backed up your patches to a card. This should fix everything for you right now.
SYNCING SPEED OF PORTAMENTO TO TEMPO [drK]
Possible work around: create a patch that uses layer switching based on either velocity or something like AT or MODWHEEL. The "normal" patch is just as it was before, but with no portamento effect. The "new" patch is enabled by the layer switch and has programmed in it an extra LFO-based sweep of the pitch. If you sync the LFO to MIDI CLOCK and KEYSYNC it it will start at the beginning each time a key is played and will run it's cycle at a time based on the current MIDI CLOCK. Set the CLOCK Divider to be the length of the slide your looking to do. Now the trick is to program in your sequence the proper control to trigger this alternative patch when you want the slide effect. If using AT or CC then just microscope-edit it in. Using velocity it is as easy as using the grid programming or step programming.
Finally, while the obvious choice for a slide is the rising (or falling ramp - you will need to create a second layer for those) also try other LFO shapes as some interesting effects may happen. Also be sure to use the LFO Offset parameter to make the effect either all positive or negative, depending on direction.
Whew!
BTW, having portamento time locked to tempo is a cool feature. Good idea!
MIXING LEVELS [Chandler]
I've seen a lot of posts about mixing boards and mixing, but I haven't seen anyone talk about mixing strategy. To many of you this stuff may seem like a no brainer, but it may help some people out.
After all one does want to get as much input level to each channel as possible, however, you do have to consider how many channels you are going to be working with including effect returns.
If everything is coming in at 99 percent of clipping/distortion and you have 8 channels of tracks & 4 tracks of effect returns, you will have a hard time keeping unity gain equal across tracks and something is going to get lost. This is why having a meter level on each channel is so helpful. The more work done up front, the easier to mix on board, and hopefully a better final mix.
I've just seen so many sound engineers in a live band set-up, crank up everything to the ninth, and then start having either feedback issues into their mains or monitors because they end up overdriving every channel. They always fall into the trap of counting on limiters or doing EQ cuts to make it all work, instead of getting the input gain set right in the first place. This is theoretically true problem for working with tunes on a mixer/PA or a recording set-up. There is only so much gain that will pass through the main out send, what you have to figure out is how much of this mix will be each channel. The art is to know the music that you are mixing to get the right combination of levels.
You can control the gain off of the instrument/amplifier, using the pre-amp gain on the board, by using the output fader on the board, and the main output faders. I highly recommend using tube microphone pre-amp or a good DI box for another level of boost or cut before your mixer. Usually the gain is more invisible using these tube boxes than the imput gain on most mixers. Compressor/Limiters also work well being put into a pre-mixer channel, it isanother way of controlling gain pre-mixing board.
Ideally, one would want to maximize the gain at the instrument/amplifier stage, use a mic w/a pre-amp or DI box to set the levels, then use the imput gain on the mixing board as a final level of control before the mixing board fader. Imput gain boosts the signal (usually) pre-fader. The channel fader controls the signal in the main mix output and usually to the effect sends (unless they are pre-fader sends, usually only found on bigger high-end boards.) Set all faders to start at 70% (usually marked by a box) on the mixer. These faders then become the final touch on the final levels. If you have to drop or rise your faders much below or above this box, your imput signal is probably too strong or weak. This is not including if you are fading something in or out, either way, it should end up in this box when at the "total" mix volume. (I get to doing an analog channel overdrive below…)
The more work you do controlling your levels before the mixer, the easier it will be to mix and usually to get a solid level going out to a PA, a multitrack recorder, or mix recorder.
A way that I use before mixing tracks or a band is to write out everything to be mixed.
Example Set-ups using a standard 8 channel board (1st four channels, 2nd four stereo channels)
505 using all outputs, assigning drum kit & a bass line to separate channels.
505 Main Sends -- go out stereo to Channels 1 & 2 on the mixer (pan 1 far Left, pan 2 far right) Drum Kit (505 Direct Send 1) - go out stereo to Channels 3 & 4 of the mixer (pan 3 far Left, pan 4 far right) Bass line (505 Direct Send 2) - go out stereo to stereo channel 5 of mixer
Reverb on effect send 1 - returning with a stereo return 1 Delay on effect send 2 - returning with a stereo return 2
OK, take in account that you have four main items. Assign a percentage of the song that should be the drums, the bassline, & the rest of the mix. Lets say that the drums are a big part, so 50% of the total mix should be the drum kit, 25% the killer bass line, & 25% the rest of the sounds mixed together. This means that on the main mixer output meter, that the initial input on these channels should add up to this percentage of what is going out to the world (i.e. your recorder, PA, etc…). OK, if you are going to have effects on these instruments you have to figure what percentage of their sound will be coming from the effect returns.
Other than checking the level on the master mix output meter on the mixer, you can sometimes solo the channel onto the meter, which shows how much of the gain that you are using on the channel. Some mixing boards and most multitrack recorders have a meter level for each channel, which is real handy for seeing where the gain is at in each channel. Ultimately, you have to leave breathing room in the main mix for each instrument. If you get the unity gain equal across the board, you will have more chances to correct the signal (i.e. you could change the level at the instrument, the pre-amp, the gain on the mixer, & the channel fader. If you have one level to the ceiling and the rest at 2 or 3, your not really hearing the instrument. Balance is the key between each stage.
Lets say I want a pretty wet drum sound with a lot of reverb, I have to then base this 50% of the total mix on a combination of the channel imput of the drums & the return reverbs on effect send 1. This goes true for the bass line and total mix with the using the reverb & delay on effect sends 1 & 2.
The best way to do mix is to set something as a starting point, the drums and base everything off of this level. If you are sending through the drums hard enough to clip the board out put, there is no headroom to add anything else. You have to have enough room in the final mix to add these other instruments to the brew. If you are wise in your initial settings, you should have plenty of room up and down to control the final mix or to add another effect. Also, if you do these precautions up front, you should not run into as much nasty clipping when using a digital mixer. If you are wanting to overdrive an analog imput, do it at the channel gain stage, then use the fader to bring the signal into the mix. (This is one time that you may find the fader not being set at the 70% box area.)
This kind of set-up is true for mixing a band or doing multi-track recording. The more you do up front, the less you have to "fix in the mix". Don't rely on EQ & compressors to bail out problems up front. Use them for sweetening in the final mix. (Although it is always a good idea to put a compressor/limiter on the insert of a mixing board, this way you don't end up blowing up someone's speakers or input on recorders.) If you are wanting to get this effect after going through some effects boxes, I would advise using the main channel inputs instead of the effect returns because usually the pre-amps are better on these channels and can handle more gain than usual line level effects returns.
This kind of pre work also comes in handy for mixing a band, a speaker, or anything using a speaker. Generally when mixing a band, I start with the drum kit, then the bass, the guitars/synths, then the vocals. You have to leave room for everything in the final mix. Although many rock band guys just never fucking get a clue that having the amps loud enough to melt metal is why they cannot hear the vocals out of their monitors. Meanwhile, the audience in the second row is having their ears burned for them are having a good time. They just don't get the fact that the guitar sound is not forming until six foot in front of the stage and the bass guitar half way across the block at these volumes… (These are problems no mixing board can solve however…) Turn the damn things down!!!! Get the tone from the amp & guitar, not the volume. It's not 1967 and you are not Hendrix trying to get over some shite bad mono PA.
Ultimately, the more you do up front, the better the result in the end. Plan ahead and don't wing it.
EXPERIMENT! [Pieter B.]
try to program as much genres you can imagine, even if you don't like house, try making a house track, you might hate it, but you can learn a lot doing it, like avoiding clichés and stuff.
SAVE OFTEN [Pieter B.]
Save the song your working on about every 15 minutes, the mc505 tends to 'paralyse' from time to time (it doesn't react to anything except turning the power off) if you push the record button really fast twice. So save if you don't want to end banging your head to the wall when it crashed for the third time in one afternoon (that happened to me some times).
INTERNAL MIDI SEND SETTINGS & PROBLEMS THAT THEY CAUSE [Christopher]
A friend and I were preparing to do a show and we decided we were going to do all new material. So we midied up the gear with the 505 as the master sending clock to his mc-303 but his machine was receiving midi data such note on of messages etc..., so I went to the midi setup pages in the 505 and disabled all the midi Tx messages including program changes. This was a mistake and as a result when I switched patterns all the program changes I had saved before did not get transmitted to my machine internally. The next time I turned the machine on to work on a pattern everything was set to patch A:001. I was saying what the ... I had written down all the patch setting so I just dialed them in but when I changed patterns all the settings I had just made carried over to the new pattern. I started checking other patterns. a wave of adrenalin rushed over my body and I was gripped by the fear.
Really, I only needed to disable this info on the Seq out (I think this is what it is called--I am at work) page for the pattern. I was freaking out and racking my brain because loads of work (8 months worth) was gone and everything in my machine was totally screwed up. I even wanted to cancel the show. So I was trying everything to fix it and track down what I might have done to cause the problem so I could escape the two days of depression I had been going through. I went back to the midi setup pages and enabled the Tx settings and Voila evrything was restored. I was relieved, to say the least, and I bought a second memory card.
I guess these Tx setting pages are used when the 505 is in sound module mode and the PC and CC's are being sent from and external Sequencer such as Logic or Cubase and you want to filter some of these messages so you can tweak stuff live or whatever. Anyway, I am pretty inexperienced with connecting MIDI gear because I don't have much. This was my first exprience connecting to another sequencer, so live and learn.
I hope this helps and you. As in every relationship when trouble arises we need to look to ourselves first. We must ask ourselves what may I have done to cause this. If we do this we can have a fuller and more meaningful relationship with our 505's.
PATTERN SAVING TIP [rEalm]
I don't know if this happens to anyone else, but sometimes when you're working on a pattern and you go to save it, it always reverts to U:A001. This is a pain, especially when everytime you have to save, it means scrolling to C:A080. I save a lot, and this can be frustrating as hell. Last night I discovered, that if after hitting WRITE, it goes to U:A001, you can then hit EXIT, and WRITE again, and it will go to the pattern you saved to last time. I don't know why it does this, but it seems to work every time. Might save some of you the hassle of having to scroll all the time like I had to.
Don't forget you can check to see if you're going to write over a previous pattern you had saved by pressing UNDO after you press WRITE. You can then toggle back and forth between the name of the current pattern you are saving, and the name of the pattern already saved to that location by pressing UNDO repeatedly. Bring on the tip wars!
FILTER SMOOTHING & KNOB SENSITIVITY [Loren]
Also... one of the 505's weak spots as we all know is the way the filter sounds when you tweak it... For smoother-sounding filter sweeps, try taking a slightly different approach.
Remember, the filter value is determined at any time by both the knob position, and the values in the filter envelope. Slowing down the envelope a little bit at certain places will "pull" the filter across the sound just as if you were sweeping it manually.. And the envelope doesn't seem to "skip" the way the knob does.. it HAS to go smoothly.... so for acid sounds, instead of twiddling the knob, try tweaking the filter envelope depth and decay..... I have come up with some VERY decent acid analog-type sounds, which if I heard somewhere else, probably would not guess came from a 505...
And I have heard many times that Roland's approach to synthesis relies a lot on EFX... well this definitely holds true for the 505.... The difference that adding a little ENHR, spectrum, compressor, etc. is AMAZING... something that sounded like a puddle of sonic slime before might come alive with shimmering vibrance with the right effect... so keep that in mind...
And another thing that always strikes me is how VERY SENSITIVE the knobs and sliders of the 505 are. It almost seems like they gave you SO much room to tweak that unless you are careful, it is too much... Especially when creating patches, working with carrier/mod structures... turning the knob just one MICRON can completely change the sound, as can any adjustment of any of the envelope values, pitch, lfo , etc. I think people overlook this great sonic range and instead interpret it as "choppy knobs"... the MC can create a LOT of sounds. Structure 1 is usually very boring.. try some of the other ones!
DRK FOLLOWS LOREN [drK]
This is a nice description and something that agrees with my own experience and readings, though I will never claim to be a "rhythm guy".
Speeding up drum tracks is really a wonderful creative tool and reveals nuances that are not obvious at the original tempo. Down-tempo-ing is also something to try. In fact one of the nicer features of the RM1x is the ability to change the clock scaling on each track. This lets you easily dial-in differing timing relationships and hear how the interact.
Another key I think to lively repeating music is in how the rhythm and the sound interact. A very nice effect is to have the rhythm and the LFOs not in sync but closely tuned. If the LFO controls some audible aspect of the sound, preferably something more subtle then the idiomatic filter sweep, then the slight drifting between the two rates will cause nice subtle movement. Never obvious when the track is only casually listened to but dynamite when it is immersed in.
Some patches naturally exhibit this effect, particularly when subjected to changes in the rate with the arpeggiator being used. This knob can be very subtle if used well and with the right patches. The "right" patch is one which uses velocity to control the sound. A trick to try is to construct a four tone patch with velocity switching (or cross-fading) between tones. If the four tones are all essentially the same but with only slight variation then your sound will "come alive" when driven by something like the arpeggiator when using the Accent control. There is a whole family of technique call "hocking" which uses differing instruments to play alternating notes in a composition. It creates the illusion of multiple instrument lines from a single passage. The above is merely a variation on this theme.
True polyrhythms can be a source of infinite fascination. In my opinion exploring these on something like the MC505 can be frustrating because of some of the "everything in sync" limitation, but it can be done. By far the easier way is to use two or more sequencing things - computer, second sequencer, arpeggiator, synth step sequencer. But don't overlook the potential of using the LFOs either! You can take a single MC505 part and just play one held note for the duration of the pattern and use the hold pedal to sustain it across the pattern repeat. With this you can then use the LFOs to program evens that do not need to be in time with the base sequence. A trivial example would be to use one LFO as a crude AMP EG (the down ramp works fine) to define a "note" event and then use the second LFO to control the pitch - maybe the S/H LFO setting? Simple "figures" even a trivial as a minor third trill can add some variety to an otherwise static sequence. Multiply this by four for each part (four tones layered) and you can get some pretty dense pseudo sequences happening. Unfortunately this does not work for percussive waveforms but you might be surprised how a decaying sawtooth controlling both the LP filter's cutoff and the AMP can be made to sound like a percussive instrument (old modular trick).
I know this is far afield from the discussion on fast tempo break-beats but really it is all very tightly related. A great deal of today's innovation in e-music has come about from challenging conventions. The above are just a sampling of a wide variety of places to explore.
LOREN WAXES ABOUT MUSIC [Loren]
A way to approach "Drill & Bass" beats on the 505 is to use the arpeggiator instead of sampled breaks. Think of it like a little sample going off every time you trigger the arpeggio.
To start, turn on the metronome, put it in real time record mode, and set the resolution to the finest setting before off (you have to put it on standby to set this).
I usually build these types of non-repeating rythms around the hihats or equivelant ticks...I usually use a fast arpeggio like 1/16 or Sequence B to lay down an initial hihat track. The oddities and breaks in this initial layer will expand out to Become the entire beat. Pay special attention to where the open hihat part goes, that is like a "relief" or "open" point in the beat, a mini pause to re-energize.
Next lay down some deep kicks... I find that Portamento B and Sequence D are excellent for solid kick patterns. The kicks should be fairly even and steady, not doing anything TOO crazy, except for at certain points (such as the 4th, 8th measure etc).
I think a Key in general to rhythm is maintaining a balance between steady, often-repeating parts and spontaneous, semi-erratic parts, and a balance between frequency ranges, "compensating" low, wide frequencies with the necessary number of ticks in the high frequency range during the same period.
Anyway, continue with either the clap or snare parts... generally the clap accents the "highest" parts, the peaks of the rhythm. It's like in pinball or super mario bros when you knock out 8 turtles and get a 1-up. After a certain combination of ticks and booms goes by, the clap has to go off to mark that spot.
Snares can also fill this role, and can also fill the offbeats, creating syncopation. In "Drill & Bass" snares typically act differently.... more like ramps in and out of different parts of the beat... like you said, with that rushing sound, they "ramp up" to drop-points in the beat, or can well up in the middle to define a "groove curve" through the beat.
For some clasic "break" snare patterns use 1/16 with the range turned to 2 octaves. Depending on which octave's snares you hit you get either the "chakachaka" of two snares, or the "ahcha ahcha" of a perfect offbeat syncopation.
Laser and blip type sounds fill in all the little spaces, hinting at all the other rhythmic variations compressed and hidden in other tempos and grooves.
I almost always come out with something that doesn't repeat at all after a round of real-time recording. I usually then cut & paste the best parts into a rhythm (again, the balance between cascade & repetition), but if you programmed several good 16 or even 32 measure rhythms, you could really get a whole song that wouldn't repeat at all or hardly at all (add a groove and you can avoid repetition with 1 BD and 1 HH!!)... A good technique is to lay down like an 8 measure hihat or hihat and bassdrum part, then copy it onto itself, making it 16 or 32 measures. Then you can have the same hihat or BD parts repeating underneath while you make nonrepeating snare and clap accents.
Experiment a lot with different arpeggios, adjusting the range, arpeggio rate, hitting different keys, multiple keys, playing little patterns, etc. and you will begin to see that you can create any rhythm with the arpeggiator. The patterns they give you are the most basic elements of rhythm that there are, and by combining the proper patterns on the proper drum notes, and add the proper groove, I truly believe that the MC-505 can generate Any Possible Rhythm within its tempo range. Of course incredibly fast and slow rhythms pulse out to infinity, but just like we only hear a certain range of vibrations, only a certain range of tempos (rhythm "wavelengths") is applicable to our type of consciousness.
However, if you think about it and look at the history and trends of music, you'll see that the range of appreciated rhythms grows as the species evolves, and that range seems to be expanding incredibly rapidly right now. Rhythms like those of Aphex Twin et al. simply DID NOT EXIST in peoples' minds say 20 years ago (except maybe for some Sufis preserving secret musical knowledge left from when they came to Earth thousands of years ago). If these "new" beats have sprung up like shrooms in the past decade, who knows what type of stuff we'll be hearing as cutting-edge in 10 or even 5 years from now?? Who knows what ANYTHING will be like 1 year from now or MONTHS at the rate things change now. :-)
HOW TO SAVE THE DELAY/REVERB/EFX ON/OFF SETTINGS [J]
So anyway when I first got my 505 all the EFX (DELAY, REV, and EFX) master buttons were ON.
A few months later they would boot OFF.
I gradually began to notice that they would cycle ON and OFF, and wondered to myself what could be causing such a change?
Well, an utterly useless (to me) tip is that the status of the master ON/OFF switches for each EFX are saved when you do a system save (go into SYSTEM/SEQUENCER/MIDI and change something and exit).
So basically if you want your 505 to boot with DELAY MASTER OFF, TURN IT OFF then go to SYSTEM and change something and EXIT (at which point it'll tell you it's saving, and to KEEP POWER ON)
FIXING YOUR 505 TO TRANSMIT MIDI DATA SMOOTHLY [Norsez]
I thought that that 505 didn't transmit MIDI data smoothly and continuously was intentional. Today I found out that this is probably a bug.
If you want your knobs to transmit MIDI data smoothly, just press shift + pad 12 for 5 times and change Local TX of the channel you want to 'B' and move a knob back and forth for a while. That will make all the knobs transmit MIDI data smoothly.
CHEAP SMARTMEDIA [John H.]
Cheap smartmedia cards at:
http://www.hardwarestreet.com/bin/catalog/getProdPage.cgi?sku=596344
or
http://www.buynowcomputers.com/welcome.asp?sku=596344
Right now the buynow price is lower, but I'm not sure how often they change or if they'll both have them in stock for long.
MIXING TIP THREAD [Erika, honey lazer, & drK]
[Erika:]
>>You´re right, but i prefer to start with drums at 100% volume and all the other sound at 50%. Then i raise the main volume knob until the desired level. And then, i make some little adjustments to the other tracks.
[honey lazer:]
> good tip. A few other little things I do- turn the volume of the entire mix down real low, and if I get 'lost' I return all the faders to the unity gain mark (that bold line/notch 3/4 of the way up a fader) and start lowering volumes of things that sound too loud. A kind of subtractive approach, where everything starts too loud and you reign them in. I do a similar thing with EQ, trying to cut frequencies rather than boost. You can often get the same effect cutting a frequency as you can boosting, and it generally sounds better that way. The whole subtractive approach helps me get away from relentlessly cranking up channels until I have no headroom and my mix is tiring on the ears.
[drK:]
The 'subtractive' approach is a solid one to use and will prevent inadvertent overloading and distortion. Especially important to do the EQ as described because a peak in the band at the wrong time can send you running for cover (and your tweeters too).
TIP FOR MONITORING LEVELS [Rob]
I find turning the volume way-down usefeul for finding what's really 'up' in the mix. With the volume down, you should only be able to hear the vocal or melody (as a general rule of course). This is not so apparent when mixing at normal or loud volumes.
HOW TO REDUCE NOISE [drK]
Soundblaster inputs = NOISE!
I *never* record through the analog ins of a sound blaster. But if that is all you have then your sort of stuck. For me when it is time to make an mp3 I import as a WAV file from the SP808. I sympathize with your problem. SoundBlaster quality really sucks big time!
Cleaning up the noise will probably not work well. My best advice is to use filtering/EQ within a program like SoundForge. Other things to do.
EQ 2 [drK]
[Responding to a query about how a filter actually knows which peaks to filter, etc.]
You may want to read some introductory material about the physics of sound. What you need to do is get comfortable about things like frequencies, harmonics, sound spectra and the like, mainly for the terminology.
OK, I'll give this a go though. Sound is made of vibrations that have a great deal of periodicity to them. These vibrations occur at a fundamental frequency (lowest component in a given sound) and range from 20-20kHz which is roughly the limits of human hearing.
A simple sound consists of a steady, non wavering single frequency. If you play one of the SINE patches on the MC505 you will get an idea how this sounds.
A more complicated sound has harmonics which are nothing more then additional frequencies that occur usually at integer multiples of the fundamental. So a complex sound at A440 pitch would contain a second harmonic at 880 Hz, a third at 1320 and so on. It is the amplitude of these harmonics, and more importantly how they change over time that gives each sound, each instrument its own color (timbre).
Listen to one of the MC505 SAW waveforms with the filter all the way open. The distinctive "buzziness" of this wave is a result of it having all harmonics. Each harmonic in the sawtooth wave is at an amplitude that is 1/n where n is its number. So the 2nd harmonic is 1/2 amplitude of the fundamental, the 3rd is 1/3 and so on. Choose now one of the SQUARE waves. Notice that it is still buzzy but has a slight hollowness to it, also it is distinct in character from the SAW. Square waves have only odd harmonics but the amplitudes follow the same relationship as the SAW - 1/3, 1/5 ... Interestingly if you take a sawtooth that is exactly twice the frequency of the first and 1/2 the amplitude and subtract it you will produce the same sound! The reason is that the harmonics when added together (via the subtraction) cancel out the even harmonics! Now listen to one of the TRI waves. Notice that it lacks some of the edge and sound a bit hollow. This is because triangle waves have a harmonic series of identical to a squarewave but the harmonics drop in amplitude as the square of the harmonic - so it goes 1/9, 1/25 ... This wave is very close in character to a pure SINE tone but still has some extra harmonic edge.
OK, that's basic waves and there spectra. Things are never this simple in nature or practice because adding motion to the harmonics is the name of the game. Let me walk you through a simple patch you can listen to. Take a sawtooth wave and run it into the lowpass filter. Adjust the filter ENV amt to zero and make the AMP env sustain at full amount with short attack and release. Play a low note (C two down from middle) note and slowly open up the filter. Since this is a lowpass filter as you increase it's cutoff frequency more of the upper harmonics of the sawtooth wave are allowed to pass which starts to restore the waves original character. Now go in the opposite direct - close the filter down. You will here the sound become more and more SINE-like until all you here is the rumbly low fundamental. If you keep adjusting the cutoff frequency more the apparent amplitude of the fundamental will even drop as it too is becoming filtered.
FILTERING IS A WAY OF REDUCING THE AMPLITUDE OF THE HARMONICS IN A WAVEFORM.
Now try increasing the resonance to almost MAX and try the same experiment. Notice as you go up that you will find places where the filter seems to cause a sound to jump out. These are the individual harmonics each being emphasized by the resonance. This can also be tried with the PEAK filter for a similar result but without the inherent low pass filtering.
Another way of looking at this is to think about light and color. Color is made up of frequencies too. The pure colors have single frequency components while the shades have multiple combinations of the primaries. If you take a good white light source (which contains equal amounts of the primaries) you can place a filter in front of it and only see the color that the filter passes. Or you can block certain primary colors and see all but that. With sound it works the same.
How does this help you do sound design or recording engineering? After a while your ear will start hear the sound not only in the totality but also, when you try, you will hear individual harmonics. You will at least be able to estimate that a given sound has some frequencies in certain bands. Armed with this you can then set about using EQ or the synth's filter to emphasis or eliminate those frequencies.
One trick I use is to take a sharp filter (like a parametric EQ peaking filter) and adjust it for its sharpest setting (high Q). I will then set its frequency close to where I think the area of interest is and then manually sweep the cutoff frequency until I find the part of the sound I am after because it will jump out (higher amplitude) when the cutoff frequency is centered on it. I can then not the setting and use this as appropriate. Even in something like the MC505 which does not have "calibrated cutoff" you can still do all of this by ear. Lets say I want to add some animation to a patch by having just the 5th harmonic modulate in and out (amplitude up and down). I would first select a Peaking filter because it will allow me to emphasis a frequency without altering the rest. next I would adjust the resonance up and the cutoff frequency until I hear the 5th harmonic. This is the setting that I want. To get it to vary with time I might use a slow LFO and have the LFO *slightly* modulate the filter cutoff frequency. with a high resonance it doesn't take much for this effect to work. of course a high amount of modulation is useful too for other purposes.
The best advice I can give is to make some basic patches and then experiment. Try to postulate what frequencies the sound has and then test your results using the high resonance sweep method.
And when your feeling like you've master that a bit try the same thing with white noise and see if you can guess why it is called "white".
EQ [drK]
You're right - EQ is just filtering.
On the conventional synth you usually have 3 basic filtering choices (four on the MC). Lowpass - used to eliminate frequencies above a certain point hipass - used to eliminate frequencies below a certain point bandpass - used to eliminate frequencies outside a certain point
Sometimes you see some other varieties notch - used to remove frequencies around a certain point (the opposite of bandpass) boost - used to accent/increase the amplitude of frequencies around a certain point
Filters have figure of merits that tell you how aggressively they cut or boost. The "order" of the filter - 2 pole, 4 pole, etc give you an indication of how sharp the filter is. In other words how quickly (in terms of frequency) the filter changes from passing to cutting. The higher the better. The rule of thumb for this is you get 6db/octave of freq change for every pole. Bandpass and notch style filters really get 1/2 that because they use a "pole" for each side of the filter. So a 2 pole lowpass (as in the MC505) gives you a cut of frequencies above the filter "center frequency" at a rate of 12db/oct.
Another aspect of a filter's performance is its "Q" or resonance. This "figure of quality" (hence "Q") is derived originally from the quality of the components that were used to make the filter (in the days of radios made with tubes!). Today it generally means two things depending if the filter is like those in a synth path or in EQ. For synths it is an indication of the gain or boost that the filter gives to those frequencies right at the filter frequency in relationship *to the rest of the "passband". The filter passband is that range of frequencies that go through unaltered. So a lowpass filter with a high Q will boost the signal's frequencies right at the cutoff more then say those frequencies below the cutoff. The higher the Q the greater the boost and the narrower the range of frequencies boosted! In the extreme at high Qs filters behave a lot like very narrow bandpass filters and can even self oscillate.
EQ is really just more of the same. The difference is that in EQ you are usually either boosting or cutting without affecting the rest of the frequencies - its a bit more discriminating. EQ can be made up of simple lowpass and high pass filters - tone controls on consumer audio is nothing more then this. Or it can have "parametric" filtering.
The biggest difference between synth filtering and EQ is what happens to the frequencies that you are not interested in. Usually in synth filters these are eliminated as best possible. In eq'ing they are left alone. Hence a lowpass EQ is really a "Shelf" filter where you can set the frequencies below the cutoff point to either be raised in amplitude (boosted) or lowered (cut). The amount of boost or cut is measured in db. The important thing is that frequencies above the cutoff in a shelf filter are left alone. Likewise for a highpass shelf filter the frequencies above the cutoff can be boosted or cut and the ones below left alone. In a way it sounds like EQ filters are backwards so this can be confusing. The bandpass EQ filter works similarly.
EQ filters also have Q but it is a bit different in meaning (though its origins are the same). In this context Q refers to how selective the filter is - how well it does its job without affecting things outside of its "band" or range of frequencies. It is a bit like filter order in that it determines the slope of the filter. The easiest thing to remember is that low Q EQ filters are a bit "softer" effecting slightly the frequencies around the cutoff more then the higher Q which are "harsher" - doing dramatic changes to the frequencies of interest without affecting those around.
The low boost on the MC505 is an example of a shelf lowpass filter. The boost setting determines how much gain the low frequencies have below the fixed cutoff of the filter.
Parametric EQ is a very versatile filtering scheme where you can set the main parameters for each of the EQ filters. This includes cutoff frequency, boost or cut amount, and the width of the region affected in terms of frequency (similar to Q). With this you can hone in on a particular range of frequencies and affect them without altering others. One "band" (filter) of EQ is generally useful for minor alterations, three very useful and five can do miracles! Often a parametric EQ will offer a lowpass shelf, a highpass shelf and one or more true parametrics with adjustable cutoff and width. The lowpass and highpass usually offer adjustable cutoff also.
EQ in a recording engineering application are usually used to surgically alter the sound of instruments and vocals so that the mix fits better together. This typically is done by cutting frequencies from one instrument or instruments in order to create frequency space for another, Sometimes it is boost that is used to accent a particular instrument's qualities. Of course eq'ing is a very creative tool also so really anything goes!
In a synth usage EQ provides a static environment for the sound. By static I mean that EQ usually does not change with time, at least in the sense that we sweep it with an envelope or LFO. (the Emu Z-plane filters however allow you to do precisely that!). It is best to think of EQ as a sort of frame for your sound. For most acoustic sounds there is a "body cavity" that has its own sonic character - peaks and valleys of frequencies. This is a very important part of accurately synthesizing something like a violin or piano. Modeling "the box" so to speak can be effectively done with parametric EQ. A surprising amount of realism can be imparted simply by creating the proper EQ. As a sound design tool overall EQ is good for creating a sense of environment (especially in conjunction with reverb). It is an important and useful tool that complements well the dynamic filters normally used in synthesizers.
Unfortunately synth manufactures by and large still view EQ as an effect as opposed to an integral part of the synthesis process. There are some exceptions. The Kurzweil VAST family, for instance, lets you use one or more blocks of EQ filtering in its mutable synth architecture. The Nord Modular (and micromodular) feature a variety of EQ modules that you can patch into your creations. But our beloved MC505 only offers EQ as a EFX.
Spend some time doing some "exercises" where your only treatments to a sound are through EQ. You'll become familiar fast and I think you will be amazed at just how much influence it can have on a sound.
NEIDHARDT'S CIRCULAR TUNING [Norsez]
Not sure if you guys are into Alternate Tuned music or not, but I have experimented with some scale tuning and thought "Neidhardt's Circular" scale sounds very nice. The following is this scale tuning on C if you wanna try that with your 505.
0 -6 -4 -4 -8 -2 -8 -2 -4 -6 -4 -8
Play your favorite chord progression on this tuning and maybe you'll find that it sounds 'transparently beautiful' like I do.
[More tunings at http://home.cybertron.com/~brtubb/018.txt]
REPLACEMENT KNOBS 2 [Guido/G-Cam]
well, to, i had nearly the same problem last year after a little gig when my reso pot was no longer where he should be... did the repair by myself
---BUT---
As the part number of pot is sent id like to add the important info that there are SEVEN different "Binding Tap tights" are in use so be very carefull to remind which sort of came from where ! I know that this can often lead to desperate situations if you dont have an exploded view from the service manual ! (I tried to order one and -with no problems- got one from Roland Hamburg, Germany)
And, using this word, you have to "explode" your darling very, very much to get into the reso-pot area (meaning to be able to solder it out and change with the new one) !
TIP one: order a service manual before or do scetches to remember wich "Binding Tap tight" was where.
TIP two: order the pot from roland, there may be little differences in pots from other providers as the lenght of stick to knob f.e.
REPLACEMENT KNOBS [Doug F.]
> I need some help with a broken knob. A friend of mine has a 505
>and a drunk friend of his tried to step over the 505 and their feet
>riped off the RESONANCE knob
It is a 10K, B taper pot. The Part Number is RK09L1140.
USE A REMOTE TO TRIGGER THE D-BEAM [Jose]
This may sound a little freaky for you, people, but
had you tried to use a TV remote control on the D-Beam ? I wondered if the infrared sensor would "feel" another source of infrared instead of the IR bouncing from your hand. I tried with my TV remote and pointed to the D-Beam in ad-lib mode and produced a repetitive triggering of 1 note.
KZOOSAX RANTS [Kzoosax]
Good People of the List,
The smoke has drifted and my mind has cleared. I would like to call on all of you to seek out your neighbor, find a line mixer & PA, and spend an evening sharing method, music, and madness. What I got out of this weekend has only begun to be expressed. Beyond meeting totally cool people, I heard one machine make four wide interpretations of its capabilities. I would venture to say that we have all only begun to scratch the surface of our potential, regardless of what great patches we've already done. Many tips were shared-- here are just a few:
1) Very simple, but relieving. When you go to write a pattern, press undo/redo after pressing write, you can scroll through your existing patterns and avoid erasing previous work. This freed me up to save even small variations as a new pattern so that I can begin to take advangtage of the MegaMix function. That's how rEalm did his whole set. (sorry to reveal your secrets, rEalm ;-)
2) Patch editing. Hadn't ever done it. I kept getting asked "is that a patch you made?" I thought it was great just to have as many new sounds in my studio as the 505 had. Only now am I starting to twist the sounds themselves and save them as my own!
3) I haven't used Step Recording mode. I tend to get a slight delay set up to the metronome, find a nice tempo, and lay down a real time cut. All my rhythm patterns sound that way. imAge tore a hole through my ears by demonstrating what you can only do with Step Recording. His shit was the bomb--tight as fast acting denture grip!
4) With regard to setting a slight delay before laying down a drum track, I found a way to tweak the 505 to give me just the right subtle effect. Set the delay on Long, turn the Feedback to between 10 and 12 o'clock. This will probably cause an obnoxious amount of delay on you drum track. Adjust the timing so that it delays to 16th notes (1 e & a, 2 e & a) Now turn to your faders, switch from "level" mode to "delay" mode, bring down the delay on the drum track, and slowly bring it up until you can hear just a hint of a nice long, but quite, delay. Fuckin' rocks! This setting has inspired me to write 5 new patterns call "Chicago Fest 1-5"
We all know what we know, but have no clue about what we don't know. (Deep, huh!) Hangin' with just three other people who use a 505 gave me insite to all sorts of methods and tips for accomplishing things I hadn't even thought of. Our listserve does a great job building a virtual bridge between us, but it can't beat the real thing. Hopefully we can schedule another Groove Fest and go even further with the event. 120,000 watt PA, outdoor concert venue, Ticketmaster (Ugh!), backstage amenities, the whole nine!!! Or maybe just at my house in Kalamazoo...
Cheers Y'all, Kzoosax
TIGHTENING UP A FINAL MIX [AB]
Another good way to tighten up a final mix is to do a 2:1 compressor across the L and R out, or if you can normalize your levels inside your computer program then compress 2:1 L/R out. This will really tighten up the mix and not falsely trigger the compressor by peak levels.
Don't worry about this slight dynamic compression because once your track is released it will get squashed to death Compressed/ Limited 10:1 by radio folks. Slight compression can also help make it so those peaks don't cause any problems when you get your acetate pressings if you choose to release on Vinyl .
I agree G-cam, reference your mixes on all types of systems, BIG & small, car, boom box, old cassette player with mono speaker built in. This referencing will definitely be worth all the trouble in the end when you are proud of your final project because it will sound great.
FIXING YOUR 505 KNOBS [?]
I stated before to fix your loose knobs with superglue, but DO NOT do this. I found a better way to fix them (with in mind that they must stay removable if some repair may be necessary!)
This is what you can do best:
This way, the knobs will sit tight and are still removable.. 8
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