Having trouble getting the mix just right!
- Thread starter Kewlpack
- Start date
Kewlpack
New member
Heheh - oh dear... just compared my song to Joe Satriani's new album... no contest! Mine's better! NOT!!! Hehehe.
How in the world does he get the volume on the tracks without clipping or fuzzing out (limiting???)???
I tried to bump the levels on a "mix down track", but the levels hit red almost immediately. What's up with that?? Maybe that's an issue...
~A~
How in the world does he get the volume on the tracks without clipping or fuzzing out (limiting???)???
I tried to bump the levels on a "mix down track", but the levels hit red almost immediately. What's up with that?? Maybe that's an issue...
~A~
Kewlpack
New member
Trying out Adobe Audition
Welp - I'm taking the original tracks and working on them in Adobe Audition... so far the sound quality is MUCH better - not to mention the whole interface is better too.
I'll try to post a better mix when I get the hang of the app.
Thanks for the great tips.
~A~
Welp - I'm taking the original tracks and working on them in Adobe Audition... so far the sound quality is MUCH better - not to mention the whole interface is better too.
I'll try to post a better mix when I get the hang of the app.
Thanks for the great tips.
~A~
JohnnyMan said:
I'm "reprinting" this so it is easier to find. By popular request I typed up a quick and dirty tutorial on better use of compression.
Here's the big secret of compression:
You should *barely* hear it working except as increasing your overall volume within the parameters you need. The average person may not even hear it working much. THAT is how the pro's set 75% of their compression, the other 25% is super squish city reserved for things like submixing drums in stereo and mixing it back in at low levels to beef stuff up.
Jim's rules for compression:
First let's define what a compressor does--which is to affect the amplitude of a signal by selectively reducing it. Compressors tend to have the following controls:
Compression ratio: this determines how 'hard' the compressor is supressing the signal. Usually described as a ratio such as 2:1, 4:1 and so forth. What this means is, after you cross the threshold setting, how many db's you have to go over to effect 1db of volume change. Thus a 4:1 ratio means that once you go over the threshold for every 4db over you will only get 1 db of amplitude change.
Threshold: this sets the decibel level that the compressor starts to work. Signal underneath the threshold will be unaffected--signal above it will be hit by the compression amount determined by the ratio. Needless to say, setting the threshold above the peaks of the signal will NOT do jack shit to the signal. You gotta set it in the path of the signal, so to speak. This is always expressed in negative db, thus a -24 threshold will compress any audio above -24db, and leave everything below it alone. (*Note, soft knee compressors start to work a bit before the threshold!)
Attack time: how long, in milliseconds, it takes for the compressor to kick in. This keeps your transient peaks unaffected and is the trick for getting a "punchy" kick or snare (the front end crack will be uncompressed and thus louder than the following signal).
Release time: once the signal falls below the threshold how long, in milliseconds, it takes for the compressor to 'let go' of the signal. For vocals and other similar instruments you want this to be fairly long like 200-250ms. For drums 75-125ms is great.
Special note on soft-knee compressors: some compressors have a soft knee function. What this does is start compressing the signal lightly as it approaches the threshold, and as you get closer to the threshold it will compress harder and harder until you reach the threshold and the full compression ratio will be utilized. This provides for fairly transparent compression and is great on vocals. Personally it sucks for drums unless you are squishing a stereo submix of drums.
Another note on stereo compressors: you should *always* link stereo sides of compressors when processing stereo signals. Once a side reaches the threshold BOTH sides get the compression. Failure to do this can lead to, for example, drums that leap in volume on one side but not the other... very assy unless that's what you really wanted. (Why god, why???)
Moving right along.....
Here are some guidelines off the top of my head:
2:1 ratio--overheads, distorted guitar, soft vocals, most synths
3:1 ratio--clamping down on overheads, acoustic guitar, most singers
4:1 ratio--bass, snare, kick drums, toms, crap singers
8:1 ratio--bad bassists, screaming vocalists, squishing the life out of stuff
12:1 ratio--out of control peaks or when you want to sound like limiting but still keep some life to it
Compression ratio and threshold are intertwined, so set both accordingly!
If you need dynamic range--LOWERthe ratio
If you need more regularity in levels--RAISE the ratio
If you just need to shave off some peaks--RAISE the threshold
If you want to affect a lot of the signal--LOWER the threshold
Here's the tricky parts that require hard decisions:
If you want more smooth sounds--LOWER attack time (under 6ms)
If you want more punch--RAISE attack time (between 7-50ms)
If you need "more" compression--LOWER attack time more
If you need "less" compression--RAISE attack time
If you need 'invisible/natural' compression--RAISE release time
If you need 'audible/percussive' compression--LOWER release time
Now pull out yer ears:
If it pumps and breaths--RAISE release time (unless you want that)
If the compression seems to disappear--LOWER release time
Finally the number one rule for compression:
ALWAYS match relative volume levels (by ear) before and after compression using makeup gain--meaning that they should be peaking about the same. If you record using my "-15 to -12dbfs with peaks no greater than -6dbfs" rule this is easy; if you tend to record sloppy and "hot" you may need compression to keep you out of the red. Don't do this to yourself.
The idea for this is that LOUDER often equates to sounding better to us, fooling us into setting duff and mookish compression settings. When dialing in compression make sure that the before and after levels are identical so you can hear the compression and not the jump in volume.
Here are some guidelines on setting makeup gain:
The lower the threshold the more makeup gain you need.
The higher the threshold the less makeup gain you need.
The higher the ratio the more makeup gain you need.
The lower the ratio the less makeup gain you need.
Further modified by:
The faster the attack the more makeup gain you can get away with.
The slower the attack time the less makeup gain you can use.
The faster the release time the less makeup gain you can use.
The slower the release time the more makeup gain you can use.
You can also calculate the amount of makeup gain you need by looking at the signals peak levels or RMS--figure this out, and then:
COMPRESSION THRESHOLD = T
SIGNAL DB PEAK = P
COMPRESSION RATIO = R
(T-P)/R = A
P-A = MAKEUP GAIN
Thus if your threshold was -24db and your signal is peaking at -12db the amount of gain being compressed is 12db total; i.e. (T-P). Divide this amount, 12db, by the ratio of 4:1--making A = 3db reduction of peaks.
Then take the signal db peak and subtract the peak reduction--in this case -12db and subtract 3db... meaning you require 9db of makeup gain to approximate the original signal level.
Last edited:
Kewlpack
New member
Ya, the last mix was a flop. 
I'm reworking things now...
The biggest issue was that I had to use headphones initially to do everything. Now I'm using my external 4.1 system to mix against and it is helping.
I also had reverb and delay on the tracks... I'm tossing most of it.
We'll see... The guitar track got "overprocessed" so I need to lay it down again. At some point.
~A~
I'm reworking things now...
The biggest issue was that I had to use headphones initially to do everything. Now I'm using my external 4.1 system to mix against and it is helping.
I also had reverb and delay on the tracks... I'm tossing most of it.
We'll see... The guitar track got "overprocessed" so I need to lay it down again. At some point.
~A~
One thing that is *ESSENTIAL* to a professional sound is mastering the parametric equalizer. Graphic equalizers just don't cut it for recording; in fact, I *NEVER* use one personally.
Parametric EQ is easier to use than you think, but it can appear tricky to master at first. It's not--just use your ears and don't over EQ.
KEEP IN MIND THIS GUIDE MAKES MORE SENSE IF YOU OPEN UP A PARAMETRIC EQ AND MESS WITH IT A LITTLE BIT WHILE YOU READ THIS!
This is a quick and dirty guide to parametric EQ.
First off, parametric EQ's have multiple bands--typically 4 or 6 bands. Each band is independent and can have its own individual settings. Most parametric EQ's have a number of MODES or FILTER TYPES available to them for each band:
HIGH PASS FILTER: will not affect freqs *higher* than the center frequency--in other words this cuts out lower frequencies (the highs PASS thru--get it?).
LOW PASS FILTER: reverse of the high pass--the freqs *lower* than the center frequency are unaffected--this cuts off high frequencies.
Both high pass and low pass filters have something called a *roll off* which may or may not be user definable; a roll off will determine the slope of how the frequencies are reduced--such as 6db per octave, 12db per octave and so forth. The greater the db reduction the more frequencies are reduced.
High and low pass filters are usually only available on the ends of the parametric EQ bands. Thus, a 4 band parametric could have a high pass filter, 2 band filters, and a low pass filter as its options.
NOTE: both low and high pass filters *ALWAYS* are used to cut frequencies--these cannot be used to boost.
SHELF FILTER (LOW OR HIGH): affects ALL frequencies from the center frequency and upwards (for high shelf filter) or below (for low shelf filter). Use carefully and sparingly. This is basically a relative of the high/low pass filters but contains no roll off.
BAND FILTER: this is the "typical" mode you will use--this will accent or cut a certain range defined by the user (see below).
After the mode, which 90% of the time you will be using a band filter type of mode, the next important thing to look at are the actual controls of the parametric eq--the center frequency, Q, and gain.
CENTER FREQUENCY: this is the epicenter of the where you are applying EQ at. Usually ranging from 20 hertz to 20,000 hertz (20 khz). This is just the center of your eq adjustment, other frequencies will be affected.
Q: this determines the width of the eq around the center frequency. The higher this number is the narrower the range. Very narrow boosts can sound "ringy" and actually go into a (bad sounding) self-oscillation due to the feedback used to create the boost.
GAIN: the "height" or "depth" of the equalization. The gain, which can be positive or negative, determines how much cut or boost you are using in that frequency range.
In general, keep all gain cuts/boosts within 6db. Most of my cuts are under 2 or 3 db's these days. If you record sounds to be *exactly* what you want you don't have to mess with them very much--resulting in a much cleaner, pro sound.
Parametric EQ is easier to use than you think, but it can appear tricky to master at first. It's not--just use your ears and don't over EQ.
KEEP IN MIND THIS GUIDE MAKES MORE SENSE IF YOU OPEN UP A PARAMETRIC EQ AND MESS WITH IT A LITTLE BIT WHILE YOU READ THIS!
This is a quick and dirty guide to parametric EQ.
First off, parametric EQ's have multiple bands--typically 4 or 6 bands. Each band is independent and can have its own individual settings. Most parametric EQ's have a number of MODES or FILTER TYPES available to them for each band:
HIGH PASS FILTER: will not affect freqs *higher* than the center frequency--in other words this cuts out lower frequencies (the highs PASS thru--get it?).
LOW PASS FILTER: reverse of the high pass--the freqs *lower* than the center frequency are unaffected--this cuts off high frequencies.
Both high pass and low pass filters have something called a *roll off* which may or may not be user definable; a roll off will determine the slope of how the frequencies are reduced--such as 6db per octave, 12db per octave and so forth. The greater the db reduction the more frequencies are reduced.
High and low pass filters are usually only available on the ends of the parametric EQ bands. Thus, a 4 band parametric could have a high pass filter, 2 band filters, and a low pass filter as its options.
NOTE: both low and high pass filters *ALWAYS* are used to cut frequencies--these cannot be used to boost.
SHELF FILTER (LOW OR HIGH): affects ALL frequencies from the center frequency and upwards (for high shelf filter) or below (for low shelf filter). Use carefully and sparingly. This is basically a relative of the high/low pass filters but contains no roll off.
BAND FILTER: this is the "typical" mode you will use--this will accent or cut a certain range defined by the user (see below).
After the mode, which 90% of the time you will be using a band filter type of mode, the next important thing to look at are the actual controls of the parametric eq--the center frequency, Q, and gain.
CENTER FREQUENCY: this is the epicenter of the where you are applying EQ at. Usually ranging from 20 hertz to 20,000 hertz (20 khz). This is just the center of your eq adjustment, other frequencies will be affected.
Q: this determines the width of the eq around the center frequency. The higher this number is the narrower the range. Very narrow boosts can sound "ringy" and actually go into a (bad sounding) self-oscillation due to the feedback used to create the boost.
GAIN: the "height" or "depth" of the equalization. The gain, which can be positive or negative, determines how much cut or boost you are using in that frequency range.
In general, keep all gain cuts/boosts within 6db. Most of my cuts are under 2 or 3 db's these days. If you record sounds to be *exactly* what you want you don't have to mess with them very much--resulting in a much cleaner, pro sound.
Once again Clone Boy - exceptional compression tutorial 
This point here is so suble & simple & obvious but makes such a huge difference in making judgements cincerning the tool being used on your audio. I have been trying to get into the habit of 'dialing the effect' up when I can (it's a little hard to do this with compression unless you keep your hand near the makeup gain - cool makeup gain post too!) as opposed to hearing the full force of the effect and dialing it down - maybe that makes more sense for eq & tube simulator type things...
Just a note on some valuable info earlier in the post - this piece here I think you have backwards...if not then I don't get it
:
At any rate great post !
This point here is so suble & simple & obvious but makes such a huge difference in making judgements cincerning the tool being used on your audio. I have been trying to get into the habit of 'dialing the effect' up when I can (it's a little hard to do this with compression unless you keep your hand near the makeup gain - cool makeup gain post too!) as opposed to hearing the full force of the effect and dialing it down - maybe that makes more sense for eq & tube simulator type things...
Cloneboy Studio said:
Just a note on some valuable info earlier in the post - this piece here I think you have backwards...if not then I don't get it
:I would think lowering the compression ratio (from 4:1 to 1.5:1 for example) would allow more dynamic range to pass thru the processor, etc...
At any rate great post !
kylen said:Just a note on some valuable info earlier in the post - this piece here I think you have backwards...if not then I don't get it
I would think lowering the compression ratio (from 4:1 to 1.5:1 for example) would allow more dynamic range to pass thru the processor, etc...
OOPS!
I corrected it in the above post. Thanks man!
Kewlpack said:
Personally I put compression on virtually every track of a song: guitars, kick, bass, vocals...
However, I tend to use *light* compression on most things. High thresholds and low ratios. Rarely do I go over 4:1, and more often than not I'm using 2:1 ratios on stuff. I want to achieve regularity but keep some of the dynamics and transients on things--I just want things a bit smoother than naturally.
Limiting should only be done during mastering IMHO.
peopleperson
I'm so sorry.
Unless Im going for an effect of some kind, I try to use compression as a problem solver with levels.
Kewlpack
New member
Tone problems discovered... VOX amp.
I know this post is more about my gear than the mix, but bear with me... I just went through a rough, sleepless week of MANY hours dealing with the following:
Note: for everyone who gets this amp (or really any other gear at all):
READ THE MANUAL 2x SLOWLY! ...and save yourself a few "duh" moments. Let me elaborate with my little story...
When I first got my VOX AD50VT, I quickly read (i.e. skimmed) through the manual to get up to speed. Okay, I thought, ready to go. Let's crank and jam! Man things sound wonderfully toneful and it's just simply fun to play and tweak and play and tweak (of course, the presets sound super the way they are). There's so much variety and the tube-tone rocks.
So, I've been recording some tracks with my two week old AD50VT via the Line Out to my PC soundcard (SoundBlaster Live series). As some of you know I posted a few versions (various mixes) of a song that was all done through the VOX via the Line Out output. At first, I was pretty happy with things.
After listening though, there were some issues with the tone on the recordings (some people mentioned it as well). I figured it was just something in my signal chain that was throwing the tone I was using into high FUZZ/OVERCOMPRESSED mode. I've literally spent about 15 hours or more on the lead track just trying to "fix" it. I've spent many more hours than that trying to balance the mix to fix the way the lead sounds with everything else (lots of learning there!). Whew - I need sleep!
After all that, I decided I'd simply dial in a better tone and redo the lead.
Yesterday, as I began to rerecord the lead track, I paid close attention to the tone. I was going through FullDrive2 Stomp > Dunlop Fasel Wah > Boss EQ > VOX > Line Out > nice PC 4.1 speaker system > PC was EQ'd fairly flat. I dialed in a preset and got started. There it was - that BUZZY tone!
Hmmm, I'll dial in another tone, something like the AC30 with medium gain (I thought I'd just try it with a semi-clean tone and see what came out). Played about two measures of the lead and the BUZZ was still there!
Even though it was colored a little differently, it was the same buzz.
I was "not a little" frustrated! I then began to do surgery on my tone-chain. I ended up just plugged straight into the amp; no EQ, no wah, no stomps; just straight Strat to Vox. I dialed in all the different amp types and whenever there was gain/overdrive present there it was... the BUZZ! (*&$!@#$) I was totally puzzled. What's the deal with this thing? It sounds awesome normally (really it does!)...
I decided to blow off any recording till I figure this out. I unplugged the cable from the Line Out on the VOX (which re-engaged the amp's speaker) and just began to "noodle" around on the fretboard. HOLY COW! The magical tone was BACK! I played a bit more... turned the dials a bit... yep, there it is. The tubey punch, cool breakup, the pick attack sensitivity, all of it.
I thought my ears were playing tricks on me. I plugged the Line Out cable back in (running to PC system) - buzzy distortion came right back and lost its soul. :shock: I A/B'd it for about 30 minutes... and anytime I was going through Line Out - the tone just broke down.
About that time, I recalled many of the pro-sound guys on the recording/mixing forums (like you guys) have talked about "never" using the Line Out on amps for recording. I didn't really pay any heed to them - until now. It's really a night-and-day difference - you have to hear it to "get it". One tone character is great (on every setting) and the other is kinda like going straight out of a fuzzbox.
Hmmm.... I figured, well let's go re-read the manual, maybe I have a setting wrong or something. As I'm reading, I come across the following (paraphrased): "...plugging into the Line Out will DISABLE the Valve Reactor Circuitry..."
.............have you ever had one of those 'Epiphany' moments?.......
Suddenly this whole thing made sense in a sick sort of "I stayed up till 2:30am everynight for nothing?!" way. The key to the Valvetronix series' tone and amazing character is the Valve Reactor Circuitry! So everytime I was pluggin' into that stupid Line Out, it was turning it off!!! To coin a popular Charlie Brown phrase... "AaaaaAAAuuuuUUuuUuuuUUGGGggghhhHHHH!"
~CAVEAT that every Valvetronix lover needs to live by~
NEVER EVER NEVER try to lay down usable tracks by using the Line Out on the back of the combos. It will kill the sweet character of the tube tone and only give you the preamp fuzz (it does change slightly with each model, but it's still fuzzy and not even close to the tone quality of the combo itself). Oh, and read the stinkin' manual a couple of times... so you don't lose precious sleep chasing a ghost!
I guess it's kinda funny...in a sick, demented sorta way.
The reason I am posting this here is because it dawned on me that all of my guitar tracks were losing distinction in the mix because everything was getting squished into the same fuzzy mess (even the acoustic stuff gets buried)... I'm betting mic'ing the amp and then the acoustic will yield better results.
Now I have to go find a Shure SM57, mic stand and adaptor to go into my PC card... Grrrr...
Submitted for your amusement.
~A~
I know this post is more about my gear than the mix, but bear with me... I just went through a rough, sleepless week of MANY hours dealing with the following:
Note: for everyone who gets this amp (or really any other gear at all):
READ THE MANUAL 2x SLOWLY! ...and save yourself a few "duh" moments. Let me elaborate with my little story...
When I first got my VOX AD50VT, I quickly read (i.e. skimmed) through the manual to get up to speed. Okay, I thought, ready to go. Let's crank and jam! Man things sound wonderfully toneful and it's just simply fun to play and tweak and play and tweak (of course, the presets sound super the way they are). There's so much variety and the tube-tone rocks.
So, I've been recording some tracks with my two week old AD50VT via the Line Out to my PC soundcard (SoundBlaster Live series). As some of you know I posted a few versions (various mixes) of a song that was all done through the VOX via the Line Out output. At first, I was pretty happy with things.
After listening though, there were some issues with the tone on the recordings (some people mentioned it as well). I figured it was just something in my signal chain that was throwing the tone I was using into high FUZZ/OVERCOMPRESSED mode. I've literally spent about 15 hours or more on the lead track just trying to "fix" it. I've spent many more hours than that trying to balance the mix to fix the way the lead sounds with everything else (lots of learning there!). Whew - I need sleep!
After all that, I decided I'd simply dial in a better tone and redo the lead.Yesterday, as I began to rerecord the lead track, I paid close attention to the tone. I was going through FullDrive2 Stomp > Dunlop Fasel Wah > Boss EQ > VOX > Line Out > nice PC 4.1 speaker system > PC was EQ'd fairly flat. I dialed in a preset and got started. There it was - that BUZZY tone!
Hmmm, I'll dial in another tone, something like the AC30 with medium gain (I thought I'd just try it with a semi-clean tone and see what came out). Played about two measures of the lead and the BUZZ was still there!
Even though it was colored a little differently, it was the same buzz. I was "not a little" frustrated!
I then began to do surgery on my tone-chain. I ended up just plugged straight into the amp; no EQ, no wah, no stomps; just straight Strat to Vox. I dialed in all the different amp types and whenever there was gain/overdrive present there it was... the BUZZ! (*&$!@#$) I was totally puzzled. What's the deal with this thing? It sounds awesome normally (really it does!)...I decided to blow off any recording till I figure this out. I unplugged the cable from the Line Out on the VOX (which re-engaged the amp's speaker) and just began to "noodle" around on the fretboard. HOLY COW!
The magical tone was BACK! I played a bit more... turned the dials a bit... yep, there it is. The tubey punch, cool breakup, the pick attack sensitivity, all of it.I thought my ears were playing tricks on me. I plugged the Line Out cable back in (running to PC system) - buzzy distortion came right back and lost its soul. :shock: I A/B'd it for about 30 minutes... and anytime I was going through Line Out - the tone just broke down.
About that time, I recalled many of the pro-sound guys on the recording/mixing forums (like you guys) have talked about "never" using the Line Out on amps for recording. I didn't really pay any heed to them - until now. It's really a night-and-day difference - you have to hear it to "get it". One tone character is great (on every setting) and the other is kinda like going straight out of a fuzzbox.
Hmmm.... I figured, well let's go re-read the manual, maybe I have a setting wrong or something. As I'm reading, I come across the following (paraphrased): "...plugging into the Line Out will DISABLE the Valve Reactor Circuitry..."
.............have you ever had one of those 'Epiphany' moments?.......
Suddenly this whole thing made sense in a sick sort of "I stayed up till 2:30am everynight for nothing?!" way. The key to the Valvetronix series' tone and amazing character is the Valve Reactor Circuitry! So everytime I was pluggin' into that stupid Line Out, it was turning it off!!! To coin a popular Charlie Brown phrase... "AaaaaAAAuuuuUUuuUuuuUUGGGggghhhHHHH!"
~CAVEAT that every Valvetronix lover needs to live by~
NEVER EVER NEVER try to lay down usable tracks by using the Line Out on the back of the combos. It will kill the sweet character of the tube tone and only give you the preamp fuzz (it does change slightly with each model, but it's still fuzzy and not even close to the tone quality of the combo itself). Oh, and read the stinkin' manual a couple of times... so you don't lose precious sleep chasing a ghost!
I guess it's kinda funny...in a sick, demented sorta way.
The reason I am posting this here is because it dawned on me that all of my guitar tracks were losing distinction in the mix because everything was getting squished into the same fuzzy mess (even the acoustic stuff gets buried)... I'm betting mic'ing the amp and then the acoustic will yield better results.
Now I have to go find a Shure SM57, mic stand and adaptor to go into my PC card... Grrrr...
Submitted for your amusement.
~A~
Last edited:
Ipso Facto
New member
Here I come to test my ear...
The lead guitar sounds like it's clipping at times.
The chorus effect on your guitar sounds like it's the 1st thing on your effect chain. *I* would move your chorus effect further down the chain (preferably after any compressor or distortion) and then increase the master volume if nescessary, to get more power. But that's just opinion...
The drums sound kind of weak. I don't know if it's a volume or compression issue or both, but something should be done.
The piano sounds panned almost totally to the right. Is that intentional? In the beggining it sounds cool IMO cause everything else comes in and balances it out but, when you come in full with the lead you could bring it back "around" (Wherever that is
).
This might be the oldest "trick" in the book(or not) but, try turning your volume down to where you can barely hear anything and then notice which instruments are dominant and which are lacking.
Oh and dude is that a real bass? If it's something derived of midi then tune down your top E string on your guitar and play the bassline on that. ANYTHING is better then a midi bass (IMO )!
The lead guitar sounds like it's clipping at times.
The chorus effect on your guitar sounds like it's the 1st thing on your effect chain. *I* would move your chorus effect further down the chain (preferably after any compressor or distortion) and then increase the master volume if nescessary, to get more power. But that's just opinion...
The drums sound kind of weak. I don't know if it's a volume or compression issue or both, but something should be done.
The piano sounds panned almost totally to the right. Is that intentional? In the beggining it sounds cool IMO cause everything else comes in and balances it out but, when you come in full with the lead you could bring it back "around" (Wherever that is
).This might be the oldest "trick" in the book(or not) but, try turning your volume down to where you can barely hear anything and then notice which instruments are dominant and which are lacking.
Oh and dude is that a real bass? If it's something derived of midi then tune down your top E string on your guitar and play the bassline on that. ANYTHING is better then a midi bass (IMO
)!
JohnnyMan
New member
Kewlpack said:Aw man... that blows... just more $$$ all the time!
Welcome to the life of a "recordist".
I started out spending $600 on a Fostex 4-tracker and a SM57 back around 1990. That was my entire studio. Now I'm thinking about spending that alone on one Shure KSM44 microphone - and its not even close to "high-end"! It just doesn't stop. Buy more gear. Buy more gear. They call it GAS (Gear acquisition syndrome).
JohnnyMan
New member
Thanks
Cloneboy,
Thank you for the very informative tutorials on compression and the parametric equalizer. I will certain use both as a reference in the future.
Addendum: I expect to benefit very much from your tutorials. I have been recording for a long time as a hobby recordist (I just record myself, mostly) but never really spent much time considering mixing. Now I can’t get enough information on the subject. I’ve been using EQ for the first time (in a specialized way) and experimenting on how to use it to separate instruments in the mix. I have always believed in compression – especially when tracking bass, vocals, and acoustic guitar but I never was too clear on what ratio to use to start out with. I do compress lightly, though. So thanks again. Any more info in these subjects would be greatly appreciated.
Another issue I have been concerned with is “what effects should be/can be applied to lead vocals”.
Another question: how do you decide reverb amounts to lead vocals and once you have made that decision how much reverb do you apply to the rest of the tracks in the mix so that everything seems to be in the same space or do we really want that? Do lead vocals usually get a plate or hall reverb and if so how wet? I have found any more than 30% and the vocal becomes too blurred. (I’m using Sonar 3 with plugins.)
Any comments or suggestions? I hope I am not being too vague.
Best regards,
John
Cloneboy,
Thank you for the very informative tutorials on compression and the parametric equalizer. I will certain use both as a reference in the future.
Addendum: I expect to benefit very much from your tutorials. I have been recording for a long time as a hobby recordist (I just record myself, mostly) but never really spent much time considering mixing. Now I can’t get enough information on the subject. I’ve been using EQ for the first time (in a specialized way) and experimenting on how to use it to separate instruments in the mix. I have always believed in compression – especially when tracking bass, vocals, and acoustic guitar but I never was too clear on what ratio to use to start out with. I do compress lightly, though. So thanks again. Any more info in these subjects would be greatly appreciated.
Another issue I have been concerned with is “what effects should be/can be applied to lead vocals”.
Another question: how do you decide reverb amounts to lead vocals and once you have made that decision how much reverb do you apply to the rest of the tracks in the mix so that everything seems to be in the same space or do we really want that? Do lead vocals usually get a plate or hall reverb and if so how wet? I have found any more than 30% and the vocal becomes too blurred. (I’m using Sonar 3 with plugins.)
Any comments or suggestions? I hope I am not being too vague.
Best regards,
John
Last edited: