|07-03-2008, 02:43 AM||#106|
|07-04-2008, 09:45 AM||#107|
thats what i used to do ......i'll probably try that.....i gotta hit chek tho cuz i tried what he said bout uploading the beats to fruity loops to work on em and the shit isnt working says theres an error......
Killbot Beats 2008 NIGGGAAAAAAAAA
Lawrlz Ya Sniffs???
"we laptop nigga, thugs on a computer." -TheShaolinAssassin
|07-08-2008, 04:33 PM||#108|
Join Date: Jul 2008
Posts: 101Rep Power: 0
Quik Techniques for EQ'n Drumz:
Techniques for EQ'n Drumz:
When it comes to equalizing your drums, don't mess around. The drums represent the heart of your mix & should be treated with respect. Below are some invaluable secrets from ModernBeats for EQ'n drums for Hip Hop:
Frequency Selection - If your serious about obtaining a clean and punchy sound for your drums, then you'll quickly learn you must be selective in the frequencies you should to boost and cut for each drum track. Most important to remember - Avoid accumulation of the same frequencies. Particularly, avoid over boosting low end frequencies. This will only muddy up your drums, as well as, mask other important tracks in the mix. Ensure your drum frequencies are even across the board!
Clean it Up - To achieve an overall punchier & cleaner mix, try cutting low end frequencies between 250 - 500hz on drum tracks such as kicks & toms. Cutting frequencies in this range will actually sharpen up drum tracks that sound too bold, harsh, & up-front. For punch, try adding 250 - 350hz into your snare & clap tracks. This will give more snap & body to the rhythm, as well as, balance out where your cut previously on your kick & tom tracks.
Adding Clarity - Cutting out the low frequencies from 100-200hz on cymbal & hi-hat tracks will add clarity while also allowing the other drum tracks that contain natural lows in those same ranges to come through in the mix. Also, boosting highs from 9khz - 12khz will add brilliance to cymbal tracks & boosting around 8khz will add crackle to your snare & clap tracks. If your not satisfied with the depth & feel on your kick tracks, then add a 4db boost at 40hz using a narrow Q of around 20. This will give the kick tracks the punch you're looking for. But ...don't forget...high pass your sub kick tracks from 34hz & below, canceling out unwanted low end rumble.
Get Your Beats Bouncin' like the Pro's:
So you've finished with the bulk of your mix - you've EQ 'd, compressed, gated, panned, & added effects to your complete satisfaction. So you're ready to lay down your final mix, right?
Now it's time to take your mix to the next level. Adding in mutes, drops, & other types of automation will help shape and give a new energy to your mix that you hadn't imagined before:
Try muting out bass & drums tracks near the end of 4 bar phrases throughout your song and you'll see what we mean. Listeners love dynamics - adding in automated mutes & drops on bass & drum tracks throughout your song adds the dynamic energy hits require. Also, experiment with different combinations of drum drops. Don't use the same exact drum drops over & over - Mix it up!
So what other types of automation will bring the final mix to higher grounds? Well, depending on your console/computer setup, all of the following can be automated: Mutes, Pans, Effects sends, EQ, Compression, Gates, and more. Don't fall short during the mix process, your beat deserves more. Imagine the energetic & dynamic possibilities with all that you could automate!
|07-08-2008, 06:20 PM||#109|
Next time stuff likes this goes here:
Props on the post though.
Some of ya'll still stuck in the 90's...
|07-08-2008, 06:45 PM||#111|
Join Date: Jul 2008
Posts: 101Rep Power: 0
my bad family i'll try to figure out the flow of where the threads go
Last edited by Jack Robin Son; 07-08-2008 at 06:48 PM.
|07-08-2008, 09:27 PM||#112|
Join Date: Jul 2008
Posts: 101Rep Power: 0
Akai Mpc Series Tips/tricks
Here is a collection of tips and tricks culled from the forums at www.mpc-tutor.com.
or everybody who samples, make sure you are keeping up with every sound you are sampling. Meaning, take the time to label the track on the Mpc with the title of the record and song number, or come up with some kind of system. I'm saying this, cause you never know when somebody might be feeling a track enough to buy it, and if you don't know, or remember where you got it from, you're basically fu#%ed, and can't use it. And yes, I'm speaking from experience!!!!!
Also, be wary of sampled sounds, you get from people, just don't use them. Ask where they got it from. Drum sounds are cool though. But like a loop, or tigh a** flute, or horn hit or something, and you know it didn't come from a keyboard, well just be careful.
If you plan on making a living from this music shi%, always make your beats with the thought that somebody may actually want to buy it. Meaning, The 1st time I walked into a big studio, to track my beats out, I was so confused and unprepared, for real. I didn't have anything labeled, I was going through all the tracks on the Mpc trying to find what tracks were being used and stuff like that. I couldn't find my kicks, snares, I was a damn mess. The engineer was looking at me like 'What the F#@K are you doing here. Hell of a learning experience. Now the engineer is like one of my best friends.
For those who don't have the 8 outs on your MPC, QUIT PLAYING! Get them bad boys. If you sell a track, you're going to need to track it out, and usually you'll have to use your own MPC. So quit buying them sacks, save that loot, and get those outs.
Hey this isnt really an advanced tip, and it may jus be common sense come to think of it but it certainly helps me and maybe it can help you:
When you got down a dope melody - or loop - (mines are usually 4-8 bars), the best way i found to create variations is like this. Lets say u got ur drums bangin hard, but u feel you can freak the samples better. What you do is mute the track with your sample. Locate the sample(s) and mess around with there params (decay, tune, etc...), and try to trim or timestretch them a little to add that spice. Next i bang out a new sample melody over the pumpin drums (this may end up being the hook, break, or wat have you).
Sure nuff the next step is to re-do your drums. Whether you do it with the original sample melody or the remixed sample memory is up to you, jus make it hott.
Next un-mute/mute those bad boys, go into step edit and add/subtract wat u want, see which ones you like, and you can paste the variety into a new sequence or wat have you.
Hope this helps, not as advanced as im sure you'll get from others, but i think this can help those starting out. peace ya'll and keep them tips comin'!!
i use track mutes for little changes and then fills and stuff for the big changes....my stuff isn't generally vocal led its more club based...so i dont tend to follow the verse-hook-verse format....so its basically the same as id do a bhip hop tune: many seqs linked by song mode... just listen to some similar tunes...you'll find most have a teaser intro, intro of main hook, emebelillshment, a breakdown and then back up for the final slam....but thats not all tunes ...percussion? its all about the sound and less can be more....ive been guilty of adding too much in the past its just a case of experimenting with say 4-5 sounds and building from there...... also house is more reliant on groove...honestly listen to some diamond hosue tracks and you'll realise there are not many elements and sometimes its just the beat for ages but cos it grooves people want it...
would use a noise gate via a PC wave editing program and play the song through the noise gate at various settings ie, threshold -40 to -20db, release/decay 100-200ms, attack 0. Play it realtime through that and maybe you will hear the drum alone somewhere in there and you can snatch it like that. If you hear it just save the noise gated wav as another name and chop that up into the peices you want, I use a native sound forge 7.0 noise gate most of the time..
OK, one is to take a string/piano, filter out the mids and highs leaving only bass for a bassline. Use that bass in Auto Chromatic (2kxl). Go into shift_6 PARAMS so you can FILTER down to zero and raise RESONANCE up to maybe 7 and it will hit/knock very hard.. Go into LOOP now and find some good start/end points along with decent zero-crossing points and turn on loop. Now go back into shift_6 PARAMS and set DECAY to maybe 50-100. Now the loop will have a release.. Like I said just a small one..
When you sample a break keep the record spinning at 45. Once in the MPC detune it so it will go back to the original speed. Makes it sound more raw.
"sample" to your old casette recorder, then play it back and resample into MPC...you get lots of tape saturation and depending on the quality of the recording unit lots of grime and dirt on your stuff.
here's loads of ways to add 'air' to your beats it's just a matter of what works for you, here's a few i still use, but this is after years of making beats, there's loads of ways out there, do some research and use your imagination!!
If you've got access to a mic and speaker, place the speaker in a room you like the sound of (hall, or tiled bathroom) and put your beat through it and record it. Experiment with mic positioning!! It also helps of you understand a little bit about how a room works. Every room has, what are know as standing waves, I don't want to go into detail but basically they are waves forms that exist in every room, and they will effect the sound in the room differently at different frequencies, creating nodes(a boost at a certain frequency) and anti-nodes (cancellation or dips at a certain frequencies) but you can use these to your advantage ie. more bass, less bass, etc..
I could write all day about this so, experiment with mic position, corners of most rooms will give more low end. For a stereo reverb effect use two mics (be weary of phase cancellation between the two mics).
Try some imaginative placement like putting the mic in a metal box, the other side of a guitar(tuning the guitar to the track you are working on). Just mess around. When you've got your reverb sample add a little bit of it back to the original beat, you'll get some sweet results and it'll get rid of that choppy sound too.
Another few techniques are turning the rec level up on the MPC and record the noise, add this to the beat.
Sample some 'air' from where ever you sampled the beats or from some where else and add it in between your individual beats.
Use an artifical delay and reverb, although I assume you don't have any by the nature of your question. So.. Copy the beat and turn it down so you can only just hear it, play it one 1/16th or 1/8th or whatever (depends on your taste) mbye do this twice or three times, each one getting lower in volume.
There's tonnes of ways to add spice to your beats, maybe these tips will get you started but try different things find your own sound. There are no rules and that's the beauty of making beats that's what it's all about. Anyone can quantise a really nice kik and snare and yes it will sound good but is that really why we started making beats? It's about adding personality to beats, put your what you stand for in every beat you make and you will hear the difference (so will every one else).
One rule I live by is that no matter how much hastle an idea is to implement or how crazy it may seem, TRY IT!!!!
I remember reading an interview with DJ Shadow a long time ago, and he said what he does is just sample some of the "air" without the actual drum attack, and use the envelope settings to fade up and fade out. Then just assign it to a pad and program it in where the digital silence is. It's sort of like using a sonic "spackle" to fill in the cracks
LOW END THEORY
Layer your loops/Sounds/drums with the same exact sample but with low end. Listen to your beat and make sure it aint distorting(adjust your velocity accordingly.) The Low end theory makes your beat sound FULL(as opposed to flat/hollow.) Just listen to all your classic hip hop albums. For me it was the Stakes is High album(the low end theory was very obvious.) Thats how they(legendary producers) do their ****...Most heads dont understand the low end theory(or even know what it is) so they try to use computers to substitute. The boom bap isnt in the computer. Its on the MPC(any of them) or SP or whatever machine your using and those dusty platters. Now using the low end theory and then importing your **** into a computer to fully master it....now your talkin good hip hop music.
"Layer your loops/Sounds/drums with the same exact sample but with low end."
Turn your Freq to like 20 and your reson up to 10....or somewhere around those numbers....keep **** with it until your loop sounds like a bass line
try taking drum loops and then FILTERING (dialing the jog dial to lower the freq. number ) until you only hear theLOW end resonating) Killer bass made this way or sample PURE BASS as in SYNEWAVES... this is the purest form of bass the 2000 and 2xl have sufficient filters to get dope **** bass from almost any drum loop
nother cool trick for yall producers if your having trouble finding a bassline to PERFECTLY and I mean PERFECTLY match your drum loop sample the same loop twice now filter down one as mentioned above till you only hear the low end now track them side by side and your bassline plays exact same as loop
you can make a bass sound by going into the sample screen and putting the the volume of your mixer up a bit then just tap on your turntable and sample the noise, filter it out, tune it and your ready to make a bassline.
you know how you might do a break using just the start and play buttons, where you make an intro or something just going back and forth between start and play, but then you're like, how do I actually program it? Try this. Take a two bar pattern that has hits on the 1. This is sequence 1. Copy it into two new sequences, 2 and 3. Make sequence 2 one bar long, but make the time signature into 2/16. Then make sequence 3 one bar long, but make the time signature 3/16. Then go to song mode and program it like this:
* sequence 2: repeat once
* sequence 3: repeat twice
* sequence 2: repeat once
* sequence 3: repeat twice
* sequence 1: repeat once
there you go, you have a hot intro to your track. It works the best when you don't have long samples in the original beat.
MORE LOW END FROM SAMPLE
I came accross a decent way to get a lot more low end out of a kick drum (the one i used was low fi (from a nintendo game) and wasn't booming enough).
* make a new copy of the kick pad
* then make this new pad play simultaniously with the original so you can hear the new sound your working with
* turn the tune down by -200 or something crazy
* bring the filter way down (maybe to 30) to get rid of the crackly high artifact it starts to produce.
* Bring the attack up since you dont want the initial punch and might want to mess with the release since it plays longer than the original
* If you want it to rumble even longer you could add slight reverb (need to mess with the settings though)
This really livened up my 8bit sample from Battletoads that just didn't have enough bass to it. It adds more low end, plus the low goes on longer than the original sample to make it sound like its booming. I also filtered the original pad down a bit to get rid of some ****.
With the MPC, there's mad options. I just want to add another way to layer. You can go and assign three different snares to three different pads. Then record your sequence with two or three of the snares hitting at the same time on their own individual tracks. From there mess with the swing settings on each of those tracks so that each snare lands in a different spot than the other. I like to use the same rim shot & record them to hit at the same time while on their own invidual track. From there I'll adjust the timing on the second rim shot track (ie: later by 1 or 2) so that it will land right after the attack of the first rim shot. This creates a thicker rim shot. Peace...
Another technique I employ is that I will actually layering my snare hits. Using the same technique as MPC-T said with stacking snares. I'll then create my basic drum pattern. Once the snare are laid in the sequence. I'll then put the snare on 16 pads and add snare hits at different pitches and/or velocities.
You can play around with this in a variety of ways. You can put one of the non-stacked snares on 16 pads and do the same thing. You can also try doing this with no quantization. You can make some very nice drum patterns like this.
Assign each snare to its own pad, then, on one pad, set 'Mode:' to SIMULT. Now you can set up to two other pads to sound whenever you hit this pad.
To blend the sounds, try shaving the attack off one, and perhaps shave some decay off another. Maybe filter the other one. Play around with these sounds until something works. Sometimes just simply layering two sounds without any changes sounds great.
If u wanna do some bangin clap snares select about 3 slightly diff claps and make sure the stronger one is center, put one to the left bout 15% and the other to the right 15%.Now program the beat and add another clap but pitch the sample down -3 and make it 2 ticks late.Do the same with anoter clap but pitch it up and move it 1 or 2 ticks forward. Now add a real clap like from the rolls royce car wash track and put it hard left then put the other one hard right. (still use the tick feature to make the sample feel lazy.
Record your HH with note repeat Time Correct [1/8(3)Triplets 53%Swing]. Just pressing the pad randomly throughout the track. Then overdub the same HH sound again but with normal 1/8 Quant and adjust swing for preference. You should have a nice HH line.
just make a beat (say in 1/16 @ 85 bpm) then open the snare track, and adjust it to 1/32, then program the hats on every other step @ 1/32, now the hats will appear to be faster (a better xample is One in a Million by aaliyah) ya know?
then go in and play with the hat track by adding parts where the sound doubles up, now where the "double ups" occur, play with the decay of the lead in hat and the attack of the lead out hat until the two transfer seamlessly to each other, presto, the funky fast hats, try and think of it like a drummer, u have only one hand, but the hat can open and close and u can strike it at any point open or closed or in between. the sound of the popular hat patterns are made by adjsuting the pitch of the hat, and the attack/decay to simulate a live drmmer striking the hat in various states of open/closed. combine these two techniques (thanks timbaland) and ur done, it takes a while, but so does programming anything complex.
yeah, what I meant by change the snare to 1/32 I meant change the HATS to 1/32, my bad. now all u gotta do is remember 1/4 note = 96 ticks
* 1/8 note = 48 ticks
* 1/16 note = 24 ticks
* 1/32 note = 12 ticks
soooo, when we open up the program and go in 1/32 every other step is also the same as a 1/16 note, ya dig, ie
1/32 = 12 ticks 2 notes in 1/32 = 24 ticks= the second one hits where it would if timing was at 1/16.
ok so if you program it at every other at 1/16 you get a tap-tap-tap-tap- sound
tap-tap-tap-tap- sound to make it double, like in Baby, when you get to near the end of the bar (in 1/16) change it up
say you end up an 188.8.131.52 bar 1 3rd beat, and u have it all done in 1/16 (tap-tap-tap-tap) now change to 1/32, and go forward from this point 1/32 note and go forward every 1/32 and add a hat, it will go like tthis nowap-tap-tap-tap-tappatappatappatappa
basically at the 184.108.40.206 point the track will change from a 1/16 to a 1/32 then back when it reaches the end, now that you have the slow to fast part needed for the cool double up effect, go back and add hat hits where u want to create the echoey part of the thing
I try and switch it up. To my understanding most lay the kicks on 1/8, snares @ 1/16 or higher, hi hiats 1/8, etc etc...
But me personally i try and mix it up here and there, im sure theres a standard as to what usually goes where, but im not too much up on that [as i dont know it]. Try different things man, dats wat music is about...hope this helps...peace
CHOPPING SAMPLE USING VELOCITY
Put a long sample on a pad. Then go to the parameters page and set the overlap to MONO. Highlight the ATTACK or DECAY value and hit OPEN WINDOW. Make these settings:
* Attack: 0
* Level: 0
Go to the main screen and set 16 LEVELS to VELOCITY using the pad you just programmed It chops the sample automatically, but not accurately. Adjusting the Level field metioned above sets the start point of the sample. Tapping pads in order 1 to 16 sounds like the sample is rewinding. Results vary depending on the sample... . but its worth checkin out.
* take a sound,
* copy it in the trim screen (section->new sound),
* put that sound on a new pad,
* set the beginning of the new sound a little bit later in the trimscreen
* pan one sound hard left, and the other hard right in the mixer.
* play both sounds at once
that's fat stereo spread, Bro. Works best with pad and string types of sound.
Duplicate sound on different pad, tune one a few cents lower. Pan hard left and right. Another fat stereo spread.
Take two bass drum samples. One with a lot of snap and high freqs.--a short, quick hit.The other, a deep, low, bass sound. Filter each to isolate their respective good qualities, snap from the first sample, bass from the second.Set each sound to trigger at the same time.(via the program) set the snappy sample to have a quick (0) attack and fast decay.Set the bassy sample to have a slower, delayed attack (3-7).This will "make room" for each of the two sounds so that they won't interfere with each other's good qualities.(time vs. amplitude)The snap will have more more headroom to sound-out since the bassier sample is silenced by the delayed attack.Once the snap decays, the bassier sound sounds-out.The trick is adjusting the cross-fade between the two sounds to shat the max. effect is heard.So they sound like one phat-**** drum.This is just the basic idea I use, mix as many samples as you like, experiment with decay times
ahem, ok all peeps here are on point in a major way, just wanna add, the way I found works best for drums, and I found out like 6 months later that dan the automator does his simmilarly, #1 layer drums, then compress 'em to bring out the meat.then layer a clean, uncompressed set of 'em over that, then layer a set with some light distortion and mix these all together, keeping the compressed sound in the middle, the distorted sound low volume and the clean drums at the highest volume.This REALLY gives drums a BOOOM sound, without being overpoweing (you know when the drums really boom, but the other sounds in the song get muddy, this allows the drum to boom without muddying up other sounds (a common problem programming drums)
This one might take some experimenting to get it how you like but here goes:
sequence your kicks on its separate track-i.e. track 3. Then copy that track to track 4. Swing track 4 using 16ths with 57% or so (depends on what you like and the original track) maybe shift 1 earlier.
You can change the velocity of this track really light to sort of get a double kick or ghost kicks. ALso you can leave it full velocity then just drop track 4 in (using mute) during a bridge or new verse or whatever, just to get a variety in your original kick track. Try less swing if you want a roll like double kick.
you can try this with the high hats as well with real light velocity.
something I've been doing... while letting an empty sequence play, Hold the Note Repeat Button while playing with a sound on the Q-Link slider. It can make some funky stuff.
Hold note repeat and press main screen, this will keep the note repeat on, enabling you to play with the note variation at the same time more easily.
SYNC PRO-TOOLS WITH AN MPC
How to sync/setup Pro Tools LE (DIGI001) and MPC2000XL via MIDI/MMC/Clock
This can be an infuriating topic because of the fact neither AKAI or Digidesign have any mention of making this type of "connection" and there are so many options and switches to configure, you sometimes never know if you're getting anywhere.
This guide will get you going fast. I'm a Hip Hop producer, and even though I charge for studio time, I don't like wasting it. And without the MPC and Pro Tools talking to each other as they should, you really are not getting your money's worth out of these things.
What you will end up with is a perfect scenario where your MPC2000XL sequences are triggered instantly when you press play (or rec+play) in Pro Tools LE. There should be no lag, space, or other problem which requires editing or pre-roll or any of that.
If all is said and done and working perfectly, you will save countless hours by just hitting rec+play and tracking your sequences a track at a time, and they will all be in time and line up perfectly in Pro Tools.
MIDI cables, Pro Tools, and the MPC2000XL, to my knowledge, do not "know" when they are hooked up; they are not like USB cables, so don't fret about that. Just follow these directions and know that this DOES work, can be replicated, and is a great way to save you hours and hours of studio time.
* 1. You only need 1 cable to make this sync trick work. Start by plugging a MIDI cable from the MIDI OUT port of your DIGI001.
* 2. Plug the other end into the MIDI IN port of your MPC. Either 1 or 2 is fine.
* 3. Hit Shift+9 on your MPC, bringing up the MIDI/Sync page.
* 4. On the left side is Sync In, set the (In:1) to correspond to the port you plugged in from step 2.
* 5. Next, Mode should be set to MIDI Clock.
* 6. Shift Early can be left at 0.
Your MPC is now ready to receive MIDI sync commands from Pro Tools. Hit the MAIN SCREEN button to return to the main MPC window. If you don't, you won't hear your sequences when triggered. I don't know why, but if you leave it on the MIDI/Sync page, it screws things up.
Pro Tools LE
* 1. Click Setups>Peripherals menu, then click the Synchronization tab
* 2. Set Device drop down to Generic MMC Reader
* 3. Set Port to DIGIDESIGN DIGI001
* 4. Minimum Sync Delay is set at 30, but this is irrelevant
* 5. UNCHECK the "Enable Control of Pro Tools LE via MMC" checkbox (if it was checked)
* 6. Click the Machine Control tab
* 7. UNCHECK the ENABLE box (if it was checked)
* 8. The other two tabs are not necessary in our example
* 9. Click OK to exit the Peripherals dialog box.
* 10. Using the Pro Tools transport window, click the Clock icon, (or use the menu Operations>Online, or hit CTRL J)
* 11. The Clock icon now flashes.
* 12. Click the MIDI menu, then click Change Tempo.
* 13. Match Pro Tools tempo with your MPC sequence tempo. If your sequence is all MIDI notes or is a loop that can be tempo adjusted without gap or spacing problems, you can change the tempo right in Pro Tools, and it will change it on the MPC all in real-time. This is very slick!
When you hit play (or rec+play) in Pro Tools, your MPC tempo should display (EXT) where the Tempo once was, but only during play. Also, you can use Pro Tools transport to FF or REW and it will move the time code display in the MPC.
What's also amazing is that if you set your Pro Tools session to bars|beats, and then click the FF or REW buttons, they play head will jog forward a bar or back a bar, and even show this right on the MPC. This is sync, baby!
These steps are not foolproof, there is often SOMETHING that can go wrong, but these are the steps I used to get it working, and my setup is basic and this step alone is now saving me hours and headaches not having to hand-edit the tracks that are off because I recorded them at-will in Pro Tools.
This is how things are supposed to be!
Enjoy, and happy MIDI synching.
PRO-TOOLS FREE AND MPC1000
Follow all the above steps except for the last few. Just enable MIDI BEAT CLOCK in the menu bar at the top and you should be sweet. Make sure MMC and MTC are not enabled! MPC1000 only accepts MIDI BEAT CLOCK.
TIGHT HOUSE MUSIC
ive found a couple of sing settings v good for house-
57% is good for that rolling house style.....good at faster tempos 125-135....
but ive found that 62% is very good for that slip slidy funky groove (think music for freaks/ carter) so i usually put the kiks and snares on the money.....maybe nudge the of hats a tick forward but basically have all my percussion and stuff on that 62% swing setting.....ill try upload a quick rhythm in a mo.....
also ive found for that straight 'billie jean' sounding beat you can keep the bds and snares on the money and the simply nudge the first hat back one tic- so lands on 49 rather than 48.....and then the second hat forward a tic- on 47....
1bd (0)-hh(49)2bd/sn(0)-hh(47) and then loop......it introduces a subtle pull-push which with the right velocity and gate times makes a nice foundation for you to build on...
bass lines are a major factor of groove- you can use the same swing settings you did for your percussion for your bass line and then go in and tweak further.....you'll find you can actually move some bass quite a way away from the original hit timings and it wont get too sloppy, moreso will open it up a bit.........
Christian Smith once told use delay with lots of feedback - it works!
Get a good compressor to make your beats pump and move
* Good ideas in the previous posts but when your trying to mix another record in you might find some elements will be out of time
* try to stick as close to the quantize as possible maybe slip a note here and there (use delay to get that real swing feel)
* Most dance music no matter what your making comes down to effects, filters, eqs and compressors to the max - really make the track talk!
there are techniquies you can use in techno that will benefit other styles... sampled house....pick your sample....could be from a salsoul or disco record...but could be anything- star dust used a tiny segment from chaka khans 'fate' looped it and worked it into a new tune....once you've picked you sample- can often be 1/2, whole bar or even a 1/4 bar long...decide how you want it to sound....you may want that classic french sound- if so rip out those filters.....mess around with dual low pass and high pass for those daft punk/cassius style filter sweeps.....or do it the american way and have the sample play and then embellish all over the top with new bass, melodies, counter melodies and thick drums.... the derrick carter remix of kid creole and the coconuts 'stool pigeon' is a good example of this....
when ive found my sample...eq'd and made some tweaked copies- ie one with a high res filter sweep, one backwards, one with no bass etc.... and load them up....get them to respond to filters and velocity, get them to cut each other off....that way you can get that akufen microhouse style...
drums: i have loads of sources...from tutors kits to vinyl samples to stuff ive layered.....with house and faster style you wnat to be able to control the length of your notes so i generally set all my percussive elements to be note off and fake a small tail so they dont cut ubruptly- you'll find if your right a house tune in 125-135 bpm and you play the whole samples tail and all the beat will get clutered and loose the groove...you need to be able to specify the note lengths so you can control the spaces in between the notes....bass drums...salsoul bass drums with 909 underneath, sample from the beginning of a house tune so its already compressed like diamonds.....
snares and claps...tight and snappy please....you can use longer ones for acents and stuff but for the tight beat you want it to snap through and end quickly...hi hats- try high passing around 2khz to 3khz....help reduce that click sound without reducing presence.....909 and 808 hats of course but try and get some other types of hi hats as they are very overused.....percussion and shakers... scoop the bass out of these....again you want to be able to control the length of these hits... bass- sub bass to under pin the root notes...layered with the bass drum for that big club sound.... bass hits from the yamaa dx range for that classic sound......proper slap/ real bass for funky bass lines....depends on the style....303 bass tends to be more techno than house...
instrumentation: anything pianos, rhodes, organs, guitars, horns, flutes, strings.....all fit as well as all manner of other stuff....nu underground mash ups incorporate hip hop accapellas.... latin and balearic incorporate many other ethnic instrumentation......its can be dark, funky, smooth and jazzy.....sequnecing: decide your tempo and swing- deeper styles have more room and tend to be slower 120-125....tech hosue can go up to 135bpm....you can use a straight swing and program your note placements....or you can use one of the mpcs swing settings for that 'tight sound'
157 is nice and loose but not too 'swung' 162 i use a lot for that skippy feel 165 is also good for that slip slidy feel.... make sure you look at your velocities....if you can nail the beat so it grooves on its own it will groove even more when you introduce some compression....bass drums always on the beat....you need to keep a tight metronome for the rest of the beat to refer to.....everything else is fair game.... shift snares back and forth for urgency/laidbackness.....shift your hihats 3 ticks forward for that urgent feel.....keep shakers in and aorund the beat but not bang on nessecarily- ditto percussion.... you want your bass drum/hihat interplay to be tight- if it is everything else if done right will flow with it....christ i can go on cant i!!!! right im gonna go now before i bore you anymore...
rules are for breaking!
I know we all want that "fat" sound but beware of adding too much compression. For those of you that sample drums from records or CDs, remember that kick did not sound like that initially. Those drums have been compressed, EQ'd, and mastered. So when you incorporate them and other samples into your mix be sure not to add any compression. This is another example of where tracking/multi-tracking comes in handy. If you solo each track you can add compression where need be instead of the whole song. Say for example you sample some drums from The Roots, but your melodies come from your sound module. The melodies may or may not need some compression, but I guarantee the drums will not. If you are already hip to this cool, if not please take note. This is something I just picked up on. When I would listen to other recordings compared to mine they would sound "thinner" when in actuality mine were too thick. This was taking away from the quality as well creating a muddiness in the kicks and bassdrums. Hope this helps atleast one person.....if so then it has served it's purpose.
use a compressor (hardware or software) to make your snares sizzle.
* turn the ratio to about 4:1
* short attack
* short release
* threshhold at about -3 to -15, (the lower the threshhold, -15 being lower, the more squashed the snare signal gets)
by "squashing" the snare you cut the peaks out of the signal, and can help avoid clipping. Raise the output from the compressor, and you will raise the perceived loudness without including dangerous clipping... hope this helps
okay swing is a real simple concept. its starts at 50% and that means any note that is recorded by the sequncer will move it to whatever timing correction note u have, the nearest one, lets say 1/16 notes, 50% means its dead on. now the more u go up the more its moving it off the 1/16th notes and sliding them. this is to make it so that the notes arent dead on and it gives it a syncopated, offbeat, feel.
say its like this this is in 8th notes 1 e 2 e 3 e 4 e now thats with 50 swing. the mpc will slide what u are playing to those notes dead on
now with say 75 swing it would be something like this 1 - e - 2 - e - 3 - e - 4 - e the dashes are where the notes are falling now. cause they have been moved half way, 75 from the timing note. say with a 65 swing it would be half of that. so it would look like 1- e- 2- e- 3- e- 4- they are slid closer.
makin it like this is the mpcs way off emulating human drum performances cause a real drummer is not on the 1 2 3 4, he has "swing" which makes it offbeat to make it sound more human and rhythmic instead of the 4 on the floor stiffness of straight quantized drums.swing can be used on anything from chops to drums to bass, iif ur **** sounds stiff and robotic try a little swing to make it seem more real
The "art" of EQ by Aaron Trumm
EQ can be used in a variety of situations, from live sound to recording to tape to mixing down. Mainly, it should be used to enhance signals that have some problem. The golden rule of EQ is less is more. If something seems fine without it, I avoid EQing it at all. Then, if I do use it, I try to remain subtle. My personal golden rule is nearly never EQ signals going to tape (as in a multitracking situation). I always try to get the original sound on tape, then I can mess with it later. Putting EQ (or any other effect) on tape usually just leads to trouble.
The other rule (the silver rule) is cutting is almost always better than boosting, especially when fixing problems. For example if a guitar sounds too thin, first try cutting high frequencies and boosting the gain a bit, instead of boosting the lows. The more clutter you can remove from a mix, the better. A better example is I very often cut a bit of high away from hats. Another example is, many times you may not hear something well in a mix...You might try cutting some frequencies in a different track that seems to be interfering, rather than boosting in the track you want to bring out. With these basic rules in mind, I''ll tell you my rules when I enter a mixdown session:
* 1. Rule Of Opposites: Usually, tracks with high sounds, (a high guitar, hats) need cutting in high frequencies and boosting in lower, and vice-versa. This is really only a starting guide, not a rule. Also, sounds that interfere with eachother can be separated in a mix by EQing them in opposite directions.
* 2. Bass usually needs a boost in the mid range somewhere and sometimes the high. This way it can cut through and be heard on smaller speakers.
* 3. Kick drums usually need that same mid and/or high boost on a subtle level so they too can cut through on smaller speakers. For hip-hop, kick needs a low end boost, but NOT TOO MUCH.
* 4. Snare drums always sound warmer with a boost in the low-mid range and some cut of the highs. An annoying CRACK can be softened with this high cut. Sometimes I boost the lows in snares to make them even fatter. But it really depends on the snare sound. The rule of opposites usually applies here. Snare sounds that were thin to begin with I usually warm up a bit, and heafty snare sounds I might thin out a bit.
* 5. Hats almost never need any EQ if they''re recorded clean. Usually an EQing for my hat tracks is to cut highs to get rid of an annoying hiss.
* 6. Guitars are simaler to snares for me. A thin original guitar might need boosting in mids and lows (depending on what the desired sound is, and what else is present in the mix) or a heafty guitar might need to be thinned out a little by cutting lows and low-mids.
* 7. Vocals usually like to have a boost in the mids or high-mids, but it depends on the voice. Vocals nearly always get lost amongst guitars...a good way to deal with this is the rule of opposites. Boost mids in the vocals and cut them in the guitar, or something similar. Vocals can also have annoying hiss or sibilance, and sometimes cutting high frequencies can help that.
* 8. Strings, and more specifically good string patches from a synth, usually need little EQ. If they are merely a support player, I may thin them out a tiny bit, or if they are meant to be present, I may thicken them in the mids a little (or sometimes the opposite...this stuff is highly subjective). But they usually work well left alone. Really clean piano or keyboard synth patches are the same way.
* 9. I like to leave reverb returns alone, but if the reverb becomes annoying and noisy, cutting some high can soften it up a bit...same with strings.
* 10. Extreme EQ setting create sounds of their own. Experiment. But for a non-novel track, be subtle.
* 11. AC hum from a track can almost always be fixed by cutting 60 Hz all the way off. (Sometimes this can take away from bass or kick sounds, but I believe that most frequencies audible in a song are above 60 Hz).
* 12. Play with EQ settings thoroughly to find appropriate settings.
* 13. I don''t mix horns too often, but when I do, I like to leave them alone. Clean horn tracks usually seem fine to me.
* 14. NEVER EVER EVER force yourself to EQ a track that sounds fine, just because you think you should use the full capabilities of the studio. NEVER NEVER NEVER!
|07-08-2008, 09:57 PM||#113|
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A Brief Introduction to Sampling Audio : A Brief Introduction To Sampling Audio
A Brief Introduction To Sampling Audio
Welcome to the next installment of Tom's Hardware Guide's do-it-yourself digital audio tutorial. The purpose of this article is to tell you all you need to know about sound sampling. The quintessential building block for most songs made in the last 15 years, there is nothing more ubiquitous than the sample. Almost every recent rock, hip hop, R&B, pop, and electronic music track now incorporates the use of samples. Wouldn't you like know how to unleash the potential of this powerful technique?
Sampling refers to digital recording and playback: taking a sound and converting it into 1's and 0's. For more on how this works, see our first article . Technically, any digital recording could be considered a sample. However, the term "sample" generally refers to a smaller piece of sound than say, a whole vocal track.
In practice, sampling is taking a snippet of sound from one source, editing it, and replaying it in a new context. It's all around you. The voice http://en.wikipedia.org/wiki/The_Voice chip of these George Bush action figures http://en.wikipedia.org/wiki/action_figure , the greeting you recorded for your voice mail http://en.wikipedia.org/wiki/Voicemail , the contents of sound effect libraries, the many modern synthesizers that use "wavetable" synthesis, and so on - samples are everywhere.
|07-08-2008, 09:59 PM||#114|
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Simple Sample Examples
Simple Sample Examples
Sampling can be used to try to represent a sound accurately, similar to how a Xerox machine can make a perfect copy. The creativity to this method lies in giving the sample new meaning, by inserting it into a different context. Sampling can also be more abstract, such as taking a sample and manipulating into something never heard before.
Say, for example, you hear a drum loop you like, such as the "amen break", which has been sampled thousands of times in electronic music, hip-hop, and even car commercials. You then could replay the sample in a loop, and put your own vocal and instrumental tracks over it. This method is a good way to add elements to a song that you wouldn't have the resources to create on your own.
This is how the notorious rap crew NWA http://en.wikipedia.org/wiki/NWA made their 1989 hit, "Straight Outta Compton." The method can make the whole much greater than the sum of the parts. However, all too many producers take a sampled melody or bass line from an old record and don't change it at all.
A classic formula for an unoriginal hip hop track would be to take an old funk record, make a loop, throw it over a 2 bar hip hop beat, and repeat for that 5 minutes. People who do this are the reason that sampling is seen as stealing, unmusical, and unoriginal. This is one of the most common, and in my opinion least extraordinary uses of sampling.
Then there is the abstract style of sampling. This involves taking a sound you like, and rather than playing it back how it sounded originally, transforming it into a new, never-heard-before sound. A common trick in this style of sampling would be to take a sample of a piano, pitch it down a couple of octaves, throw a little reverb on, and...Voila! You have a cool bass sound.
But this, too, has been done a million times before. Step outside the box http://en.wikipedia.org/wiki/Outside_The_Box ! What happens if you sample the sound of yourself opening a bottle of soda and slow it down to create a cool percussive sound? What if you sample a piece of paper being torn and play it in reverse? More on this later.
|07-08-2008, 10:00 PM||#115|
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Sampling In Hardware Or Software?
Sampling In Hardware Or Software?
Now that you're thinking of all kinds of cool things you could sample, let's get down to the nitty gritty... how are you going to sample? Traditionally, using a dedicated hardware sampler was the only reasonable choice. In the last several years, however, advances in computing power have given rise to the popular use of software based samplers http://en.wikipedia.org/wiki/sampler .
Obviously both have their advantages, but I prefer the flexibility of software samplers. A soft sampler's storage capacity will only be limited by your hard disk space, it has no knobs to break, requires no cables, is on your computer screen rather than a small LCD, and takes up no desk space.
Some of the commonly used soft samplers include Native Instrument's Kontakt and Intakt, Tascam's Gigastudio, Logic's EXS24, and Propellerhead's Recycle http://en.wikipedia.org/wiki/ReCycle . All of these programs have their merits, and pitfalls. Many DAWs (Digital Audio Workstations) come with a sampler built in, and for most purposes these will suffice. Ableton Live http://en.wikipedia.org/wiki/Ableton_Live has the "Simpler", Logic has the "EXS-24", and Reason has the "NN-XT" and "NN-19".
Of course, hardware samplers are far from obsolete. One big advantage of hardware samplers is that they will never get a virus, and will probably never crash. Although I have seen musicians playing live with soft samplers, hardware samplers offer a peace of mind that no computer can offer.
In addition to their stability and cool looks, each hardware sampler offers a distinct sound and feel that can only be approximated by soft samplers. An example of a classic hardware sampler is the EMU SP-1200. This 12-bit drum sampler only has 10 seconds of sample time, but its warm, low fidelity http://en.wikipedia.org/wiki/Fidelity sound makes it coveted by many producers. Another amazing sampler is the AKAI MPC 2000, which has velocity sensitive pads that can be played as a musical instrument.
Although there are some really cool hardware samplers available, most of them are cumbersome, don't allow the user to undo changes, and editing is hard on a small LCD screen http://en.wikipedia.org/wiki/Liquid_crystal_display - if you're even lucky enough to have a screen. Overall, I'd recommend beginners stay away from hardware samplers.
|07-08-2008, 10:02 PM||#116|
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Be Picky With Samples
Be Picky With Samples
Now that you have a sampler, you need material to sample. Although, you can take a sample from any CD, tape, radio show, phone call, and so forth, you must be careful to not infringe upon anyone's copyrighted material. One good way to do this is to make sure that your samples http://en.wikipedia.org/wiki/Sample are completely unrecognizable. If nobody can tell what it is... nobody can sue you!
An alternative to using material that has a copyright license is the Creative Common License. Creative Commons licensed material often can be sampled legally. Or, safest of all, be totally legit and sample your own material.
Most soft samplers actually don't record so much as they edit and play back samples. So to actually record a sample into your computer, you will need to record into a DAW http://en.wikipedia.org/wiki/Daw . Once you have your sample saved, its time to fire up your sampler and let the fun begin.
The logical first step is going to be editing. Editing a sample can be a complex process, but for the purposes of this article, we'll only be picking a starting point and an end point.
Select a view of the waveform of your sample, and zoom in until you can see the points where each individual wave crosses http://en.wikipedia.org/wiki/Crosses form a positive to a negative value. This point, called the zero crossing point, is the exact point where you want to cut. If you cut in the middle of one of the waves (on top of the hill-shaped lines of the waveform), your sample will sound clipped.
See the illustration above, noting the position of the green bar, which indicates the starting point of the sample in each example. Remember, your ears are the best judge of whether your start and end points work or not.
If your sample is going to be played in a continuous loop, you'll generally want your loop to be a multiple of two, assuming that your time signature is 4/4. This is just a guideline for more conventional compositions. You can get some complex polyrhythms by having loops of different lengths playing together. For example, a four bar loop and a five bar loop playing simultaneously will line up every twenty beats if they are both at the same tempo. This of course is just a basic example and you can experiment with much stranger loop lengths.
|07-08-2008, 10:03 PM||#117|
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Stretching, Multisampling, Velocity Switching, Filters
Now that you have recorded and edited your sample, the fun can begin. The first thing you'll want to do is map your sample to the keyboard, a process usually called - you guessed it - keymapping. You may notice that stretching http://en.wikipedia.org/wiki/Stretching a sample across the whole keyboard sounds pretty goofy, which can be cool. However, if you want your sounds to be realistic, you'll have to record the same sound at many different pitches, creating a "multisample." As you can imagine, playing a different sample every octave will sound good, and every 5th will sound even better, and so on. Some instruments will require more samples than others to sound realistic, but be careful not to go overboard.
If you want to sound even more realistic, you could record different dynamics of the sound you are sampling. This process, called velocity http://en.wikipedia.org/wiki/Velocity switching, triggers different samples based on the velocity value of the midi signal triggering it. However, this concept is more difficult, and will not be discussed further in this article. It's just good for you to know that it exists.
Now that you have your basic sound, you can begin to manipulate it; there really are endless possibilities. For example, you could take a guitar http://en.wikipedia.org/wiki/Guitar sound, and pass it through a low pass filter with a cutoff at around 800 Hz. Then on another track, you could take the same sample and put it through a resonant high pass filter at around 2500 Hz. Maybe apply a bit of a flanger effect to the second sample, and you now have a unique sound.
What if you played the sample in reverse? Many soft samplers now come with built-in effects, which usually require much less processor power than using a dedicated plugin for that effect. Feeling lo-fi? You could resample your new creation at a lower sample rate and bit depth. How would it sound at 22 KHz / 8 bit? The only way to find out is experiment, so get to it!
"Looking at a sampler the way it was used first - to try to simulate real instruments - you didn't have to get a session guitarist and you could just be like, 'Hey, I can have an orchestra in my track, and I can have a guitar, and it sounds real!' And I think that's the wrong way to use sampling.
The right way is to get the guitar, and go, 'Right, that's a guitar. Let's make it into something that a guitar could never possibly be.' Take it away from the source and try to make it something else. Might as well just get a bloody guitarist if you want a guitarist. There's plenty of them." - Amon Tobin
This quote speaks volumes. Although sampling can be a practical way to incorporate an ordinary element like drums or a bass line into your composition, it can also be a versatile tool for exploring new sonic horizons. Have fun - and don't get sued.
|07-09-2008, 10:21 AM||#118|
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How To Use a Compressor
How To Use a Compressor
People are frequently asking how to use a compressor. Here is a little article I have written to explain how you go about it.
Most effects processors are fairly simple to use; plug in an equaliser (for example), twiddle the controls, and listen to the output, and you pretty much know what you're doing, and all you need is some experience behind you.
Compressors don't fall into this category. Plug them in and listen. What's it doing? Unless someone has told you, then you probably won't know. Play with the controls. What do they do? Don't know either. What do the indicators mean? Difficult to tell. It's all a bit frustrating really...
Unfortunately you need to be *told* what a compressor does. Furthermore - even after you know what it does - someone needs to explain why the things that it does are considered useful. You won't figure it out for yourself.
Normally - for non-technical people - the explanations of what a compressor does, are so bewildering that they end up even more confused than they were before: "xDBs in, equals yDBs out, over zDb threshold, according to this graph" etc. etc.
Fortunately, I have a friend who explains it very well, and very succinctly:
"What does a compressor do, Alan?"
"It makes the loud bits quieter."
"I see... But surely if it just makes the loud bits quieter, can't you then turn EVERYTHING right up, and make get everything really, REALLY loud?"
So there you go. Simple isn't it? A compressor just makes the loud bits quieter, allowing you to crank everything up to maximum volume. But under what situations would this be useful?
Firstly, there's the obvious application of making your CDs sound as loud as possible. This trend is starting to get a bit silly, and is beginning to prevent people from producing albums of good dynamic range. Nevertheless, if you master a rock or pop album with no compression at all, then the chances are it will sound pitifully quiet compared to the rest of the CDs in peoples record collections. It will probably sound like it has been severely under-recorded. Compression lets you get a much higher average level onto the CD without affecting the music too much.
A very practical application of compression is in live PA setups such as a rock concert. There is a danger that very, very loud sounds will blow up the loudspeakers as well as risking serious hearing damage for those near the stage. The solution is to put a compressor in place. This makes the loud bits - and in this case only the very loudest of the loud bits - quieter, so as to avoid damage to equipment (and people). Such hard compression of only the very top peaks of music, is called "limiting", and is so useful that many compressors have a "limiting" function in addition to normal compresson, and so you will have to read the manual if you want to use this feature in addition to using the compressor for additional "normal" compression.
Another example is that many vocalists have poor microphone technique. When they sing quietly, they sing *far* too quietly. When they sing loudly, they are *way* too loud. A compressor can reduce the "dynamic range" of the vocalist to a more managable level, which is why a compressor is sometimes called a "Dynamics Processor".
Bass guitar is another instrument which can be hard to play consistently throughout a song. Any minor errors in the bass guitarists playing can leave "holes" in the song where the bass seems to disappear. A compressor can help keep the performance at a consistent level.
There are obviously many other instruments and sounds that could benefit from compression at some time.
So, as you can see, there are many applications for a compressor. Basically, in any situation where sound volume levels are getting out of control, a compressor can be used to "tame" the extremes of volume and keep it within a reasonable range, entirely according to your needs. Obviously a very powerful tool.
Using one, on the other hand, is not so simple...
Why? Because a high-quality compressor, with "good" settings, is designed so that you can't "hear" it working as such - so it's difficult to tell when it is doing the right thing. There have been red-faced moments for many engineers - including myself - when they have spent many minutes carefully adjusting the settings, proudly declaring them "perfect", and then realise that the compressor is in fact, switched off. Very embarrasing.
There is also the added problem that the markings on the controls of many compressors are not very accurate. They are only there for a "guide" - and to help you restore settings later, so simply looking at the controls may not be a good indication of what the compressor is actually up to.
Many compressors - including software plugins - don't even have meters on them, and needless to say, this makes it incredibly difficult to know if they are operating correctly.
Because compressors have many different applications, the way that you use a compressor depends very much on what you are trying to achieve with it. In this article, we will look at four main applications of a compressor which are all quite different. Most applications are just variations on these four different uses, so they should serve as a good starting point for most of the things you will want to do.
The four main applications that we will look at, are:
1. Hard limiting - to prevent speakers or digital recordings from overload
2. Compressing an instrument or vocal
3. Adding "punch" to bass drums and bass guitars
4. Compressing a final mix
In addition, we will look at a specialised fifth example:
1. De-essing a sibilant vocal
But before we get into these, let's look at the theory behind compressors and what the controls actually do. This is a little difficult to understand at first, so don't worry if you haven't "got it" the first time around. It will make more sense after you've experimented a bit with a real compressor in front of you. Note that not all compressors have all of these controls, and some compressors are very "minimalist" indeed. If you don't have all these controls, then look at the compressors instruction manual to see what preset values the "missing" controls are set to.
What The Controls Do
Firstly, in order to compress the volume range of something into a more "workable" volume range, you need to have in your mind an idea of what the lowest "normal" volume level is, and what the "loudest" volume level is, and have a mental idea of how "loud" you are prepared to let the loudest get.
The "Threshold" control, sets the volume level at which the compressor starts to do its work. Below this volume level, the compressor will literally do absolutely nothing. So you basically set the "Threshold" control to the lowest volume level at which you want the compressor to start working. We will discuss in the examples how you actually make this setting. Naturally, if the "Threshold" control is set to maximum, the compressor won't ever do anything at all because the level of the music is usually way below this level, and therefore remains totally unnaffected.
The "Ratio" control sets how "powerful" the compressor is. At its lowest setting (1:1), the compressor literally does nothing, and is effectively "switched off". On the other hand, at its highest setting (normally marked 20:1 or even infinity-to-one), the compressor is 100% powerful - so powerful in fact that it TOTALLY PREVENTS the volume level getting even the *slightest* bit louder than the threshold level! Hard to believe? Try it and see. Set the compressor ratio at maximum, play some sound through the compressor and start turning the threshold level down until you hear the effect. If you are playing solo drums through the compressor the effect is quite astounding.
The only problem with doing this is that (naturally) the total volume gets so much quieter, because you are "constraining" it (compressing it) - so very much. That's why compressors are almost always equipped with a powerful gain control marked "Output" or "Gain make-up" in order to boost the volume level back up to a reasonable level after it has been "squashed" down.
Every time you turn the "Threshold" down, you are "constraining" the sound more and more, and making it quieter, and so you almost always need to use the "Output" control to boost the level back up again. This is a bit irritating, so several compressors have a switch - normally marked something like "Auto gain make-up" or similar - to automatically boost the output as you turn the "Threshold" down. It's not on every compressor, but it is a nice little feature to have, and saves you fiddling about with the "Output" control all the time. For clarity in the following examples though, I have assumed you either don't have this switch, or that it is turned off.
So far so good. "Threshold", "Ratio", and "Output" are the main controls on an compressor, and "theoretically" give you everything you need. So what are the other controls for?
Well, sometimes - in the real world - things aren't quite so simple. For example, you can have a vocal that is sometimes too quiet, sometimes too loud, and occasionally, way, way, way, way too loud. Wouldn't it be nice if the compressor somehow had an automatic "Ratio" control?
That's why many compressors have a "soft-knee" or "over-easy" control. With the "soft-knee" control turned on, the compressor doesn't simply and immediately "kick-in" at the level set by the "Threshold" control - it merely "starts" to work. As the level gets louder and louder, it reaches a level where it is finally reaching the "power" of compression that is set by the "Ratio" control.
So if you wanted to control a vocal that was wildy out-of-control in terms of levels, you could switch on the "soft-knee" control, set a "Ratio" much higher than normal, and set the "Threshold" control to the quietest "acceptable" vocal sound level. When the vocal exceeds this level with the "Soft-knee" control switched on, the compressor starts to compress at fairly moderate levels. If however, the vocal gets wildly out of control and attempts to get *seriously* loud, then the compressor starts working much harder to pull it back to sensible levels. It's a bit like having an automatic "Ratio" control, with the maximum compression "power" controlled by the setting of the "Ratio" knob on the front panel.
Then there are the "Attack" and "Release" controls. So what do these do?
If you've followed this explanation so far, you'll realise that a compressor is a bit like having a smart guy hanging onto a volume control and adjusting it by hand according to the music. But how quickly can this "person" respond? Well, the "Attack" control, adjusts how quickly this "person" is, at turning down the volume when things get too loud. The "Release" control is how quickly that same "person" can turn the volume back up again when things have calmed down.
But why would you want to adjust this? Surely you would want it to be instantaneous? (after all, it *is* supposed to by an automatic system...)
It turns out that in practice, in many situations, you don't want the volume to be "instantly" cranked down the moment things get too loud. Under certain conditions you can really *hear* the volume being pulled down, and this is very undesireable. Instead, it *sometimes* sounds better if the "person" is a bit sloppy and slow at yanking the volume down. The "Attack" control affects this sloppyness.
What about the "Release" control? Well, in a similar way, if the compressor is too fast at turning the volume control back up again, you can hear it working (the audible effect is known as "pumping"). It just sounds "artificial". So the "Release" control adjusts the speed at which the compressor "recovers" after yanking down the volume. The exact speed which sounds "correct" depends on the music, so that's why you can adjust it by hand. The examples following in a moment give some suggested settings, but by all means experiment in order to find the most "natural" sounding setting.
And that leads us to another control. It is a switch, and it is sometimes marked "Automatic" and sometimes marked "Peak/RMS". So what does this switch do?
Well, as I mentioned before, the "Attack" and "Release" settings really depend on the music you are using the compressor on. But music continually changes. What the "Automatic" or "Peak/RMS" switch does, is to switch on an automatic setting that attempts to "listen" to the music and continually set the correct "Attack" and "Release" settings for you. Of course it doesn't always do the best job, and that is why you also have manual control if you want it. It is important to realise that with this switch turned on, the "Attack" and "Release" controls are disabled and will do nothing. Some compressors (unfortunately) don't have "Attack" and "Release" controls at all, and are either set to preset values, or permanently set to RMS (automatic).
There is also an IN/OUT switch (often marked BYPASS). This is essential. It is there so you can switch the compressing action on and off and thus hear the difference your changes have made. To make best use of this switch, you need to set the "Output" control such that the sound appears to be at roughly the same level irrespective of whether the compressor is switched IN or OUT - this allows you to easilly make comparisons by listening.
The final control on most stereo or two-channel compressors is called "Link". What is this for?
Whenever you adjust the volume control on a stereo mix, you always expect both left and right volume levels to change at the same time don't you? (otherwise the mix would wander off to one side or the other). That's what the "Link" control is for. It makes sure that both left and right hand volumes always change in time with each other, so the mix stays "in the middle".
As an added bonus, the "Link" control *usually* (although, not always) disables one set of compressor controls on a two-channel unit, and takes all of its settings from just one set of controls. This is because on a stereo signal you normally want *exactly* the same settings on both sides - as well as keeping the volume levels equal. This is not the case on all compressors though, so it is important that you check your manual to find out whether you need to set the controls on both channels to be the same, or whether you only need to use one set of controls (the other ones being disabled).
On some compressors, even if one set of controls is disabled, the "Output" controls for each side may be independant (don't ask me why - it does seem a bit silly) - again, you *must* check the manual, as it is not always easy to tell simply by playing with the settings and listening.
That completes our "tour" of the controls. I hope you understood it. Read it a couple of more times if you don't, and if you're still feeling lost, perhaps it might come to you after you've tried these examples.
So: Now onto the examples...
Example 1: Hard Limiting
The problem: You are doing a live gig or an important digital recording. You want to leave the music completely untouched, but what you don't want, is totally unexpected loud peaks causing damage or distortion.
The solution: You want to stop TOTALLY any music or sound exceeding your expected maximum level. This is an emergency situation! But this is also quite simple to do with a compressor. Set the "Attack" and "Release" controls to their fastest - after all, it will only "kick-in" during cases of emergency, and you want it to respond immediately (to prevent distortion), and you also want it to recover immediately (so no-one notices anything happened). Make sure that any "Automatic" or "Peak/RMS" switch is turned off - so that the "Attack" and "Release" controls actually work and are not in automatic mode. The "over-easy" or "soft-knee" switch (if present) should be turned off too.
Then set the "Threshold" control to maximum (probably marked +10Db or +20Db, but on some digital plugins it may be marked as zero). This will prevent the compressor doing anything just yet.
Then set the "Ratio" control to maximum (normally marked 20:1 or even infinity-to-one). You won't hear anything happening just yet, because the "Threshold" control is set to maximum - effectively bypassing the unit.
Now, play the LOUDEST MUSIC SIGNAL YOU EVER EXPECT TO HEAR through the compressor, and look at the levels.
Now, slowly turn down the "Threshold" control, carefully listening and looking at the levels. The moment you even *start* to hear a decrease in volume or see it on the meters then stop, and back off a tiny bit. You have found your optimum settings.
Just to check, try playing some excessively loud music through the compressor. You will find that it refuses to exceed the maximum level you have set, no matter how loud the input!
Needless to say, if you get a bit silly and try to blast the compressor with INCREDIBLY loud music, you may indeed hear the compressor start to distort (it still won't exceed that maximum level though!). But this is just unrealistic. You are setting it up to handle only the most unusual, unexpected and extreme cases, which will be well below the level of distortion.
Example 2: Compression of an Instrument or Vocal
The problem: You are working with a fairly good vocalist. Normally when they sing loudly everything is fine - but every now and again they sing their little heart out so much that the recording either distorts, or is simply just way too loud. Unfortunately the vocalist is so unpredicatable that when this happens you don't have time to adjust the recording levels because it happens almost at random, and is difficult to predict.
The solution: An ideal application for a compressor! Start with a good recording level for normal recording, and with a fast "Attack" and a moderately slow "Release" on the compressor (ensuring these controls are on a "manual" setting). Also switch ON the "Over-easy" or "Soft-knee" button (if the compressor has one).
As before, begin with both the "Threshold" and "Ratio" at maximum. Whilst the vocalist is singing at a fairly QUIET to moderate level, slowly turn down the "Threshold" control until either your ears or the meters detect the slightest faint drop in level. If your compressor has a "Gain reduction" meter it should *just* begin to indicate a change. Turn up the "Output" control until the quiet part is at a good level for you.
Now go to the LOUDEST part of the song and get the vocalist to sing (or play back a recording). With the ratio at maximum, you should now find that - ironically - the sound is far too quiet! Simply turn the ratio control down until the level is just about as loud as seems reasonable. The "Gain reduction" meter - if you have one - will probably be lighting up lots of pretty lamps at this point (unless it is just a boring moving pointer :-)
As a final pass on this example, you might want to get the artist to perform the song once through (or playback a "take" if you are compressing on playback), and at this point you might want to play with the "Attack" or "Release" settings to get the most "natural" sound. Be careful if choosing an slow "Attack" though, as it might allow the compressor to "overshoot" and exceed the levels which you so carefully set previously.
Example 3: Adding "Punch" to a Sound
(normally bass instruments like bass guitar or bass drum)
The problem: The artist is performing a fairly rhythmic pattern, but somehow they don't seem to be "punching" through the mix, even though their sound is basically quite good. Every time they play a riff you know you really want to "feel" the "impact" - but it is simply not there.
The solution: Although compressors are normally associated with *reducing* peak levels, did you know that they are capable of actually GENERATING amazing peaks?
This technique generally only works with "percussive" sounding instruments like drums, guitar (including bass), and spiky keyboard sounds like "clavinet", that are playing a rhythmically "pulsating" part.
The technique relies on the fact that the "Attack" control can be used to make the compressor respond in a sloppy way - thereby allowing lound signals to "overshoot" and generate peaks that weren't even there in the first place!
To do this, start with a moderately slow release, a SLOW attack, and with the ratio and threshold at maximum. The "Soft-knee" or "Over-easy" control (if present) should be OFF.
Play back the quietest part of the performance, and as before, turn down the "Threshold" gradually. You should find a setting where although the instrument is starting to get a bit quieter, it is starting to have more "punch" to it. Use the "Output" control to restore the level to a good volume.
Now go to the loudest part of the song. You will find at these settings that the instrument is - surpringly - too quiet. Turn down the ratio until the sound is loud enough.
Now check out the quiet part of the song again. You might now find that it is not as punchy anymore, and you might have to turn down the "Threshold" some more (and of course boost the "Output" to compensate).
Finally, rehearse the part (or playback), and adjust the "Attack" to give you the "punchiness" you need overall.
The "Release" control is quite critical in this scenario too. If you have it set too fast, you can hear the compressor "breathing" or "pumping" (you'll know what I mean when you hear it!). On the other hand, if you set "Release" too slow, then you will start to lose the "punchiness" - it is a tricky balance.
Example 4: Compressing a Final Mix
Ooooh! This is the trickiest one of the bunch! You will probably have one of two problems. Either (a) the mix overall doesn't sound "punchy" enough - which requires slightly different settings to the previous example, OR - (b) you have the more common problem - you simply can't get your mix to sound "loud" enough compared to other recordings that you have in your collection.
The problem (a): The mix overall doesn't sound "punchy" enough
The solution (a): If your mix doesn't sound punchy enough you have to start with some "preset" settings on your compressor as follows:
Start with the "Automatic" or "Peak/RMS" switch turned ON (RMS setting). Music is a complex thing, and a "final mix" even more so. The "Automatic" or "RMS" setting will literally "listen" to your music and try to find the "ideal" settings for both the "Attack" and "Release" controls and disable them. If your compressor doesn't have an "Automatic" or "RMS" setting, then set both the "Attack" and "Release" settings to medium. In both cases we will end up adjusting them manually later so don't worry too much.
Set (as before), the "Threshold" setting to maximum (which "bypasses" the compressor), but this time pre-set the "Ratio" control to about 3:1 or thereabouts. Now, whilst playing the mix gradually turn down the "Threshold" level until you start to get a more punchy sound. You will (as always) have to turn up the output to compensate.
When you can hear the compressor making a difference, try experimenting with the "Attack" and "Release" settings. If you previously set "RMS" or "Automatic" ON, then try to match both "Attack" and "Release" to the same sound as "Automatic" and use that as your starting point. The slower the "Attack" the longer the overshoot. Sometimes a short attack will sound good (making quick transients), other times, a slower attack will sound more appropriate.
It's a good idea to go around ALL the controls in turn, making slight changes until you believe that you have the best settings on all of them. Use the IN/OUT button to compare results with the original - using the "Output" control to match the sound level between the IN and OUT settings, so they are at the same volume - this greatly helps make a good comparison.
The problem (b): You can't get your mix to sound "loud" enough compared to other recordings
The solution (b): You really need TWO sorts of compression here. Firstly, you need "limiting" set up as per example (1) previously. Turn down the "Threshold" until you can start to hear the limiter making an unpleasant difference to the mix. Then turn it back up a bit, and try to find the position where you have the best balance between cutting down the peaks, and making an undesireable change to the music. In most cases it should be possible to apply quite a lot of limiting without any significant difference to the sound of the track.
Now that you have trimmed off the peaks, you can crank up the "Output" to a much more respectable level for mastering on CD. But you might want your CD to sound louder still. If that's the case, then apply another compressor BEFORE the limiter and just try some conventional compression as in solution (b) above (but probably with a faster attack). Many mastering compressors have a compressor AND a limiter combined in one unit for this very purpose.
As an alternative approach, set quite a fierce compression (5:1 or more), and switch on the "over-easy" or "soft-knee" button, and with fast attack. Start (as always) with the "Threshold" high, and slowly turn it down until you achieve the balance between a good amount of compression, and best sound quality. Adjust the "Release" control to help minimise how much you can "hear" the compressor working. The speed of "Release" setting is different depending on the speed and type of music - let your ears judge it.
That concludes our four main examples.
Most analog compressors have a "Side-chain" socket on the back, and now some plugins are offering a similar facility too. So what's it for?
The compressor works by feeding the sound through the compressor itself, but also by feeding the sound to the compressor "control system". The control system "listens" to the sound and controls the compressor volume.
The side-chain is a system that lets you insert something - like a graphic equaliser for example - immediately before the compressors control system. Note this is NOT in the main audio path and doesn't affect the sound as such - just the way the compressor responds. This system lets you over-emphasise a certain frequency that you want the compressor to listen out for. For example:
Example 5: De-essing a Vocal
The problem: You have a vocal, but sounds such as the letters "S" and "T" are sounding really harsh, and burst through the mix too much. You don't want to equalise them out of the vocal sound, because the vocal sound is actually quite nice apart from those explosive "S" and "T" sounds.
The solution: Using fast "Attack" and quite fast "Release", set the compressor at about 3:1 and then place an equaliser into the side-chain. Suck out all bottom end, and middle, and apply a boost at around 3 to 6KHz. You will find that adjusting the "Threshold" will control how powerful the "S" and "T" sounds can get. Overdo it and the vocal can sound very strange. Set it so that it gets the "S" and "T" sounds just how you like them.
Note: It is understandable to think that you might need two compressors on a vocal - one to perform de-essing, followed by another one doing "normal" compression. This is not so. One compressor can do the two jobs at the same time! Simply compress the vocal as normal. Then insert an equaliser into the side chain, and apply a small boost around the sibilant region (around 3KHz-6KHz). This will cause the compressor to over-react on sibilant sounds, thereby de-essing at the same time as compressing. Use the EQ BOOST to control how much sibilant sounds are CUT.
It is VERY, VERY IMPORTANT not to overdo de-essing. If you do, the singer will sound like they have a lisp. Or should that be lithp? :-) In any case, once you've screwed it up by overdoing it during recording, then there is little you can do to rescue it later - so go easy on it! (you can always de-ess some more during mixing if required).
It's important to understand that the equaliser you choose to place in the side-chain does NOT affect the frequency response of the sound going THROUGH the compressor, just what the compressor internally LISTENS to. It therefore effects how the overall volume responds to changes in volume at certain frequencies. This is known as "Frequency SENSITIVE compression". It is also possible to purchase more complex compressors that actually DO affect the frequency response in different bands, and this is known as "Frequency SELECTIVE" compression - there is a big difference between the two, although the names are similar, and even professionals get the two terms mixed up sometimes.
The example above isn't the only use for the side-chain - and indeed you don't even have to feed the same sound into the side-chain as you are feeding into the main input. You could for example feed the sound of your voice into the side-chain. In that way you can create a system that automatically fades down music when you speak, and fades it back up when you stop speaking - the same feature that some disco consoles have - except you have full control over the fade in/out rates using the "Attack" and "Release" controls.
In a similar way, you could feed the reverb returns from your lead vocal reverb through the compressor, but plug the dry vocal into the side-chain. That gives you a system where the reverb fades down when the vocalist sings, giving quite a "dry" sound, but returns to a strong "wet" reverb inbetween words and phrases in the song. It keeps the reverb from messing up the vocal in the parts when the words get busy. Such a technique is also useful for controlling the level of repeat "echo" effects at the end of phrases.
In all of the above examples, the settings and approaches suggested are merely a guide. Your best teacher of compression is your own ears, and the compressors that you own. When you find a setting that really works on a certain instrument, write it down - it will save you a lot of time later when you next record that same instrument. You need to work with your compressor for a long time, and develop a good working relationship with it, until you can really trust what it is up to.
Every model of compressor on sale sounds different. That's why people talk with great affection about certain old valve compressors, or perhaps a particular model of DBX compressor that they love (DBX are the world leaders in sound compression harware, and make by far most of the chips in the world that achieve it. This means that many compressors not even made by DBX, often contain DBX chips at the center inside them).
Settings vary between equipment. A setting that sounds great on one compressor, often sounds terrible on another. This applies to software plugins too, which is surprising, as one would expect the maths and figures to be identical in each one.
The fact that people have a personal preference for different types of compressed sound, means that there will always be a market for compressors from different manufacturers. There will always be the "classic" compressors that almost everyone likes, and there will also be a number of obscure quirky units and plugins that only appeal to a select few.
Compression is an extremely difficult thing to describe in writing, and you really need to hear compression - in all its different forms - to get an understanding of how it can help you. Never apply compression to something simply because other people do. Apply it because you KNOW that you really NEED it and that you UNDERSTAND exactly what it is DOING to the sound. If in doubt, compress too little rather than too much (it is very difficult - indeed, often impossible - to undo bad compression later), and remember that too little compression during recording can always be made up for when mixing later (at the expense of a little quality).
As always, practice makes perfect, and I hope that this article has gone some way towards demystifying the process for you.
|07-09-2008, 10:29 AM||#119|
Join Date: Jul 2008
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Compression "The Breakdown to Sound"
The term compression has several meanings in audio - there is lossy data compression (e.g. MP3), lossless data compression (a wave file compressed using a Zip program), and level compression and/or limiting. In this article, I concentrate on the last form, although a few words about MP3 are in order (later).
I recently read a wonderful article by a mastering engineer by the name of Bob Katz (see references, below). Bob was adamant that many producers, engineers and musicians have joined a new race to see who's CD can be the "hottest" (i.e. loudest). There is a mistaken belief that this makes the sound more exciting - it doesn't - it makes it boring, and very tedious to listen to ...
Just like commercials on TV, which are compressed to within an inch of their lives. Does anyone find the sound satisfying? I'd be very surprised to hear someone (other than the producer) say yes - they are annoying, seem to be much louder than the programme you were enjoying, and cause a great many people to hit the mute button on the remote the instant they start (I'm one of them ).
Compression Versus Limiting
So what is the difference between these two effects? Depending on your outlook, either not much or a great deal. A limiter is usually set to a fixed threshold, and any signal that attempts to exceed the threshold is pulled back (attenuated) by exactly that amount needed to maintain the predetermined level. If the input gain is set way high, then all signals below the threshold (including noise) are boosted - in the extreme case so everything is the same volume (again including noise!). Limiters are "hard" compressors - the absolute level is fixed, and the compression ratio may be as high as 100:1 or more. This means that the input signal must increase by 100 "units" to make the output increase by one unit. Many limiters claim that the ultimate compression ratio is infinity, however this is probably an over estimation of the true figure.
A compressor uses much the same (or at least similar) circuitry as a limiter. While some compressors boost the level of signals below a preset threshold by a predetermined amount and reduce the level of signals above the same threshold, this type of compressor is most commonly used in noise reduction systems - an expander is used at the playback end to return the original dynamic range.
The majority of compressors use a threshold setting (like a limiter), and reduce the gain progressively once this is exceeded. Compression ratios of perhaps 2:1 are common, so the output will rise (or fall) by one unit for every two units of input change. A 50dB dynamic range (above threshold) is therefore reduced to 25dB (from softest to loudest signal). Unlike limiting, the compression threshold is typically set lower than the peak level - the actual threshold level could be anything from +8dB to -40dB, depending on the effect desired.
For example, a guitar (after suitable amplification) may produce transients of perhaps 5V peak, but yield an average level of only 500mV. This represents 10:1 or 10dB peak to average ratio. The peaks are produced as the pick strikes (or releases if you prefer) the strings, and the average is predominantly the normal decay of the note before the next is played. The VU meter (I refer here to real ones, not the stupid things you so often see that bear no resemblance to a proper VU meter) gives a good indication of the average level, and therefore the perceived loudness (VU = Volume Unit).
A PPM (Peak Programme Meter) shows the peak levels - no surprise there. Some meters, mainly electronic versions, provide both indications. A bar shows the average (VU) level, and a dot that "sticks" at some higher level shows the peak amplitude.
By using compression, the same guitar may have the maximum level reduced to perhaps 1V - the average level will now be higher as well (softer sounds are amplified, loud ones attenuated). Peak to average ratio may be reduced to 6dB or less, and the note will seem to just hang on forever ... well not quite, but you get the idea.
This sort of compression is common on percussion, strings, vocals - in fact almost anything. It is appropriate if (and only if) it provides the sound the artist wants - when compression is used just to make something sound louder, then it is better to just turn up the volume. This way, the original dynamics are preserved. Incorrectly used, compressor/limiters will flatten the sound, and remove the life and soul of the music. IMO, compressors are incorrectly used in the vast majority of modern recordings.
Compressor/limiters are usually fairly complex electronically. Since they already have a voltage controlled amplifier (VCA) circuit that must be of the highest quality to satisfy audio professionals, this can be used for other things as well, with little real increase in cost or complexity (they are already complex, so a little more won't hurt :-)
A common addition to these audio tools is a noise gate. This is provided with compressor/limiters, and is used to gate (or switch off) any signal below a preset minimum. Noise gates are used to remove unwanted low level signals, but are sometimes used to mess up the sound completely by removing the ambience. Better a little noise and a complete sound than a quietly decaying ambience that suddenly just stops. Used properly, a noise gate can seem to eliminate background hiss completely, while letting the signal through (the hiss is still there, but you can't hear it when the signal is present). Used improperly, the initial parts of sounds are cut off, and the natural decay is not present. This is (fortunately) rare in pro studios.
One final feature offered in many units is a "de-esser". The sibilants ("sssss" sounds) in vocals are often over emphasised by close microphone placement, mic characteristics, the vocalist, or equalisation - and often a combination of these. This can be very unpleasant, so the de-esser does exactly what its name implies - it reduces the sibilant sounds by an amount that the recording engineer can set according to need (or taste)
Why Use Compression?
Compressors and limiters are used in music for a multitude of reasons. The first (and should be the only) reason is for the sound. Used properly, a compressor - or more correctly a limiter - will place an absolute cap on the maximum level that can be passed. This is invaluable for preventing a large PA system from distorting, or making certain that the ADC (Analogue to Digital Converter) does not clip (exceed the maximum conversion voltage). Digital distortion is extremely unpleasant, and is to be avoided, as with all forms of hard clipping.
There are many other reasons to use compression or limiting. Many instruments do not have the sustain that the musician desires, and this can be corrected by using a compressor to extend the note. As the signal fades, the compressor increases its gain, so the note lasts longer.
Another reason is to restrict the dynamic range. Movie soundtracks are a prime example. If the maximum level of a car bomb exploding or a shotgun fired at close range were to be reproduced, and all conversations were at the normal level, no-one in the theatre would hear anything that was said, or would be deafened instantly by the explosions. By reducing the dynamic range, both can be accommodated at levels that are appropriate, but limited to an acceptable maximum and minimum loudness.
By contrast, many trailers and theatre advertisements are heavily compressed - they have a consistent loudness that is greater than that of the main feature. This technique can work for a limited period (it gets your attention), but becomes very tiring very quickly.
The over use of compression results in a flat, lifeless reproduction. In his article, Bob Katz refers to "wimpy loud sound", and at some stage we've all heard it. You put on a CD, and it is LOUD, so you turn it down, so the loud parts don't leave you loudspeaker cones on the floor. You wait, you listen, you wait some more ... it never happens! There are no loud sections! There are no quiet sections. Everything is at the same volume from beginning to end, and the result is indeed wimpy. Certainly the CD is louder than others you own, but it is the same volume from start to finish and leaves you as flat as the sound.
How to make music bereft of life - compress it until it bleeds (to death). No hi-fi, regardless of cost or sophistication can make rubbish like that sound good, since there is virtually nothing that can be done. An expander (essentially the opposite of a compressor) may restore some vestige of what the artist intended, but compression is not easily undone unless you can obtain the exact reverse of the original settings - this is both difficult and time consuming to even attempt, assuming that you have the equipment in the first place (very few hi-fi systems incorporate an expander, so most of us are well and truly screwed).
Compression is commonly used in the final mix, and this is where things can go seriously wrong - everything is at the same volume, peak to average ratio is minimal, and the resulting sound is almost always worse than it was before the compression was applied. Used correctly, a small amount of compression may be useful with some musical styles, but it is completely unsuited to others. I have several CDs that sound "exciting" at first, but the sameness of having a constant barrage of sound at the same level becomes extremely fatiguing in only a short time. On some, I can hear the compressor/limiter acting ("breathing" or "pumping" are terms commonly used for this effect), which means that it has been over used, and the CD is then relegated to the "don't bother listening to this" pile. Most unfortunately, this pile is getting bigger, and many of the modern CDs are worse than older ones because of the stupid, unnecessary and pointless game of one-upmanship by the record companies, all trying to get the "hottest" CD on the block.
I don't want it to be "hot", I want it to sound the way it should, with real dynamics, soft and loud passages, and things that make me jump! Fortunately, I am not alone, but unfortunately, record companies are still producing material for people with crap systems - "Make it sound good on a crap system - we'll sell more". This is rubbish - people with only a boom box don't care that much, otherwise they would strive for something better. Those among us with good or excellent systems should not have to listen to something that was mixed on a pair of near field monitors with the quality of a transistor radio, and compressed so heavily that it has lost all of the dynamics that make music what it is - or should be!
If the CD is a little quieter than expected, then I simply turn up the volume - even without a remote control, this is hardly an arduous task. Better that than have a whole pile of "hot" CDs that I can't bear to listen to because they have had all their life removed by an over zealous compressor-head.
Peak to Average Ratio & Dynamic Range
All music has a peak to average ratio (and dynamic range - see below), since there are peaks and dips in the level (even when heavily compressed), and the average level must be lower than the peaks. The trick is to know what the peak to average ratio should be. It is commonly quoted as being between 10dB and 20dB (a power ratio of between 10:1 and 100:1). By this reasoning, music with a 10dB P-A ratio will need perhaps 50W to handle the peaks, but will provide an average power of only 5W. This is typical of a lot of music, and even some orchestral music will be at this ratio without any compression (relatively uncommon, but Bob Katz has experienced exactly this).
A ratio of 20dB is at the other extreme, so the same 50W amplifier will only produce an average power of 0.5W - this is where the use of high powered amplifiers for hi-fi is important - by the time the average power is high enough, the peak power is massive. A quick example ...
You want to listen to music at 90dB (SPL). Your speakers are rated at (say for convenience) 90dB/m/W, so with two of them, the effective sound pressure (at 1 metre) is 93dB SPL with 1W into each channel. You (the listener) are some distance away, so the level may be 3 to 12dB lower at the listening position. We shall assume 6dB as a reasonable guess for a typical listening room (although it may be considerably more than that, depending on room treatment, furnishings, etc.).
For 1W per channel, your SPL will be about 87dB SPL, so to get the extra 3dB, the power must be doubled to 2W per channel. If you have music with a peak to average ratio of 20dB, you will need 200W per channel to reproduce the music without distortion - assuming that you have that much power, the peak SPL will be in the order of 110dB ...
(90dB + 20dB P-A ratio = seriously loud).
Compression is your friend! Such a high P-A ratio will cause most high end systems (and their owner's ears) grief at high levels, so some degree of compression will make the reproduction less arduous on your system (and a lot less likely to frighten the cat to the point where it perches tightly on your head :-) The magic is to find the ratio that keeps the ratio to a reasonable figure (and there are no absolutes here!), while preserving the soul of the music. It can be done, and I have many CDs and vinyl albums that do it very well (Bob will also tell anyone who cares to listen how to do it well, too).
It can also be done very badly, and so many new releases do just that. Mind you, a lot of old releases were just as bad - this is not a new phenomenon, but has been happening ever since the compressor was first invented - or at least used in anger.
Is there a difference between peak to average ratio and dynamic range? The answer depends entirely how long the averaging period is. Generally, the two are considered separate, but with enough compression they become equivalent. The dynamic range of a piece of music is the difference between the softest passage and the loudest - it takes little imagination to realise that if it is sufficiently heavily compressed there will be no difference at all.
With a good mix, and a compressor/limiter that is correctly adjusted, the difference between the two will be less than in "real life", but great enough to create excitement - this is where experience and careful adjustment come in. The range of sounds we can hear (and will be assaulted by) is enormous, from the faint rustling of leaves on a very quiet night through to jack-hammers or jet planes (and certain motorcycles!). To expect to reproduce this range from a home hi-fi or theatre system is generally not possible and is undesirable anyway.
The difference between the two is blurred - there is an almost infinite grey-scale, rather than any black and white distinction between the two, so I shall leave it at that.
A discussion of compressor/limiters would be incomplete without a brief explanation of the features and controls typically offered. A typical unit may offer the following (adapted from real specs for a typical unit) ...
Max. Input Level: 10V RMS (+22dBu)
Dynamic Range: 118dB
Signal to Noise Ratio: >100dB
Headroom: 18dB <0.05% at +4dBu, with 6dB compression
Distortion: <0.05% at +4dBu, with 6dB compression
Limiting Threshold: -40dBu to +20dBu
Attack Time: 0.1ms - 200ms
Release Time: 50ms - 3s
Compression Ratio: 1:1 - 100:1 with selectable hard or soft compression knee
Some of these terms are self-explanatory, while others may need a little more information.
• Maximum Input Level
This is the maximum signal (in Volts RMS and/or dBm or dBu **). If this is exceeded, the input stage will distort due to clipping. This indicates that a unit with the above specification cannot be used for a direct speaker feed (few can, and a Direct Injection box is nearly always needed).
• Dynamic Range
The range of the quietest sound (signal) to the loudest. The spec above is misleading, since it ranges from the noise floor to the maximum input level. The effective dynamic range is limited by the minimum signal to noise ratio the engineer will accept, so subtracting (say) 50dB from the above would be reasonable.
• Signal to Noise Ratio
The level of noise referred to a specified output level. Again, the output level is not specified above, so we don't know if the actual noise floor is better than -100dBu or -78dBu, since the reference input or output level was not defined. Many manufacturers use A-Weighting for these measurements, which is also misleading for most applications. A-Weighting should be used only where the sound source is some distance from the listener (possibly hundreds of metres), or is expected never to exceed about 70dB SPL near field.
This can be tricky, since there are many different ways to describe it. Most commonly, headroom is the difference (in dB) between the typical or specified output level, and the maximum the unit can provide without distortion.
Another can of worms! Distortion measurements are not really meaningful if the level and frequency are not stated. It is even harder with anything that uses a VCA, since their distortion can vary considerably depending on the degree of amplification or attenuation. Distortion (probably) should be worst case - the frequency, amplitude and amount of gain that produces the most distortion. This would probably look bad in the specs, so is not used.
• Limiting Threshold
This is the range of signal level where the unit can be set to function. Below the threshold, a limiter does nothing - it just passes the signal straight through (with maybe a little fixed gain or loss, as appropriate). Above the threshold, signals are attenuated by an amount determined by the compression ratio (see below).
• Attack Time
A measure of how long it takes before the unit changes the gain when a signal is applied. This may be very fast to prevent anything from exceeding the threshold, or deliberately slowed down to allow a percussive attack to the instrument. A relatively slow attack might be used with drums to increase the apparent dynamics, whilst actually reducing them.
• Release Time
The release time determines how long the gain takes to return to "normal" after it has been reduced. A short release squashes the dynamics of the sound completely, and a very long release time ensures that all material remains at a constant fixed peak level, while still allowing normal variations. The overall dynamics (of a musical piece) are still compressed, since the soft passages will allow the unit to apply the maximum gain. This is all controllable.
• Compression Ratio
As discussed above, this can be low (1:1 means no compression), almost infinite, or anywhere in between. An infinite (or high) ratio is a limiter, anything between 1:1 and about 10:1 is a compressor, although some would argue (correctly) that 10:1 is really limiting. Some units also offer a "soft" or "hard" knee (the threshold). A soft knee means that the onset of compression or limiting is gradual, ranging over a few dB, while a hard knee means that once the threshold is reached, the limiting action is immediate, with no gradual onset.
** dBm is a reference level based on 1mW into 600 ohms. This represents a voltage of about 775mV. dBu is based in a reference level of 1V RMS (also sometimes seen as dBV).
What To Listen For
An excellent way to hear compressor/limiters in action is an outside broadcast on TV. While the presenter is speaking, the level is constant, with very little variation - even when they are off-axis from the microphone. When there is a break in the commentary, the background noise can be heard to increase at a (relatively) fixed rate, until it is as loud as the presenter's voice, or someone starts speaking again. It is then instantly reduced to where it was before.
As you get used to what to listen for, you will hear many CDs where the level of the backing track falls when the singer (or someone else) starts making their noises - this (to many audio professionals) is quite normal, but it is not - it is a typical case of over application of compression in the final mix. When an additional sound is added to those already present, it is supposed to get louder - this is called dynamics (or even "micro-dynamics" - a reduced scale version of the real thing).
These effects are especially noticeable with commercials on radio or TV - listen for them so the sound can be identified. Should you purchase a CD that does the same, complain to the record company - they have ruined your music!
What About Lossy Data Compression?
MP3 - love it or hate it, it is here (probably) to stay. As can be determined from a multitude of sources, MPEG Layer 3 (or MP3 for short) discards information that theory (and a lot of experimental testing) indicated would be inaudible. It uses a well known characteristic of our hearing called "masking", where it is known (and can be proven) that certain frequencies and levels are completely inaudible when accompanied by another signal at a higher level. The points where masking take effect are beyond the scope of this article, but differ according to relative levels and frequency, and the frequency band itself. An MP3 encoder breaks the signals into sub-bands using filters, and each is treated differently according to a set of rules built into the encoder signal processor.
While it is generally considered that a high bit rate (128kb/s or above) MP3 track is of "near CD quality", many people will dispute this vehemently. My own experiments and listening tests indicate that imaging is poor, and the precise placement of instruments and vocals is missing. Some instruments - especially the harpsichord - sound completely different when encoded, almost regardless of the bit rate. A good test is to "rip" some pink noise (preferably generated by an analogue source), and compare the difference.
There should be no difference - or at least it should be inaudible, but this is not the case! At 320kb/s the difference is barely audible - one has to listen carefully to hear it, or figure out just what to listen for ... but there is a difference, and it also shows up very clearly on the analyser of Winamp. The peaks are flattened, so the dynamic range (or peak to average ratio) is degraded, and the sound of the noise lacks "life" compared to the original recording.
If we can hear a difference with noise, why would music be any different? It isn't! Ok, noise has a relatively constant bandwidth (DC to daylight in the extreme), and excites all frequencies more or less simultaneously. Well, so does a lot of music, albeit for short periods at a time.
Will my comments here make MP3 go away? Of course not, and nor should it go away, because it is a useful way to archive recordings, or provide people (who insist on not hearing approaching traffic while they run or cycle) a convenient medium for portable sound.
Even Worse Things That Can Happen
On a final note (pardon the pun :-) a reader recently sent me an out-take of a CD. It was clipped! Not just compressed and limited to the maximum (that too), but with actual clipping - flat tops on some peaks. I asked him to check his setup very carefully to ensure that the record level was not set too high, and he assured me that he had at least 3dB of headroom above the peak CD level.
At some stage, I shall check some of the CDs I have that annoy me because of the constant loudness to see if they have the same problem. Probably not, since the clipped CD was from an "indie" (independent producer) so would not have had the controls in place one would expect from an established mastering house.
Be that as it may, there is not really much point in setting up the "ultimate" hi-fi system, with headroom to spare and almost zero distortions of any kind, only to have the music CD pre-distorted, compressed, limited and bent so far out of shape that it is no longer useful for anything other than a coaster.