POST your RIG !

the only real way to blend your drums better so the track doesn't feel "isolated" is to have all the stems of a given jam and do a proper mix and the rev is especially a crucial point ! usually different rev can be used in a mix but a "general" rev bus receiving each stem (or almost any of them) helps ALOT to unify all the seps in a sort of team pack ... also the dynamic treatment on the master (eq, multi band comp ...) is usually the last phase to polish a mix which will unify the stems :)

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@ Nilton,
Awesome, I have yet to physically see a Fractal unit, and hear one live.
Most people I rub elbows with are poor.. LOL..
Just kidding, will be listening to your tracks closer now..
Thanks for sharing.

@ Mpointon,
I am going to say, your on the right track , and I have said previously I love your sound.
Also have read Oliv's response to your question.

We haven't touched on the master track of things. Was hoping to kinda make our way there, for future listeners / watchers. And nubies alike. To learn as a input to output flow.

I agree with Oliv on most points. They are --
a. The Multi-comp.. A MUST HAVE in any setup.. The Multi-comp does 2 things, maybe more.
One the comp. part. You get this, but it goes further. Sure it sets the ceiling, and floor.
The multi part is, 4 separate comps, each for a different freq. set. Sub, mid, mid-hi, highs.
You can solo each one and dial it in. Then overall the comp. is better because you haven't sacrificed any freq.'s for another to shine.
b. the master treatment.
This is an art, nothing scientific about it for me. Yes I have certain stuff deployed, but where is just as important as what. Meaning, I have the Apogee dithering tool at the end of the chain in slot 8. Where it is POST master fader. And the Multi-comp resides in the slot just above that in 7, where it is also POST master fader. This allows me to adjust what the pre-amps are going to get, signal-wise. NOT input voltage-wise. They do intertwine, but, mostly the mix of low's and high's.
What I didn't totally agree with is one reverb type for alot of tracks.
No offense Oliv. :)
Just for me, I like to mix the drums by sound, meaning, the cymbals will have a reverb type, the toms will have one, the kick has it's own, and the snare has it's own.
I believe that control of the reverb sound to the instrument is easier, and sounds better for the same reason you would use a multi-comp.
Controlling different frequencies with a different room, and slap, or sustain.
Because, the speed of travel for the sound is different, even to the same wall 10 meters away.
Hi-hat will reach it faster than the kick, etc. etc.

Thanks for the input as always M.

@ OliVBee,

Thanks for joining in the convo.
All views are interesting to me. Different worlds and thoughts.
I was curious what you use on your acoustic guitar tracks.
Is that an Fhole mic, or multi-mic, Fhole, and fretboard ?
They sound great, and are very dynamic.

~MK

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So...
That brings us to " OK, I have found the sound ! " Let's record.

1. Do you tend to let the dynamics of the track happen ?
Meaning do you dial up the gain on the record track, to where it sounds good, or do you go by the input gain meter, all the way to -.3 or closer to zero ?
(Clipping) in the digital world.
I do the latter.

2. Do you color your track with the effects sound, or record clean only.
Or record both at once, so you can tweak later.
I do the latter.

3. If you re-amp, do you find your playing to be on time, or just a little different ? Because, the effect you had during recording, made you have timing differences ?
Ex. longer sustain because of effect. etc.

I don't find a lot of differences, but then I might be tuning the new amp to "FIT" to what I had been doing in the other effect. Ex. tuning in the delay, so it's closer to the 1st amp. etc.

4. Mic and /or amp, free-air, and live with the results, like in the old days ?

I would, in the case of the song being acoustic, and that is the the biggest one.
Or, certain types of amp sounds, old marshall plexi, old vox. etc.

Again, computing all the variables, the room, the mic, placement, humidity, etc.
Can you re-produce this every time ? Or does it become a conundrum ?

I ask this because, once someone becomes happy with their sound, they want to use it.
They want to re-create it, and record it again. At least the ones I have spoken to.
Sometimes they want a new sound, but 60 percent want to re-use so to speak.

That is the input part.

Next will be Mixing / effects.

~MK

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When recording digitally things are quite different from analog.
1) Noise floor is much lower
2) Clippning is much nastier
So i stay well away from any clipping level and normalize all tracs after recording.
The axefx has the possibility to send both the input and the processed signal over usb simultaneously and reamp as well but i havent tried that yet.

The axefx has IR cab/mike simulation, so that takes the hassle out question 4. Everything is as you want it to be but muck clearer and totally reproducable. And Ir opens up some new possibilities. Any linear response can be captured. I have tried playing a fretless bass through a ir from a acoutic double bass. It sounded quite convincing.. And it opens up for new comments like: Where did i put my double bass? Wait a second there it is!

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@ Nilton,
Thanks for the reply, and I see what you mean about IR.
That is real cool in the way of re-use, and the reproduction aspect.
Again, I really need to get my hands on an AxeFx unit to try it.
I probably will want to buy one.. haha..
And if it can do a dry mix, and wet mix at the same time, way cool.

Do you find yourself in a world of tweaking more than playing ?
Or.
Has the AxeFX brought the time of playing up and the amount of tweaking down ?

I would have to say the POD HD is a lot of tweaking. But, once you get the sound, it stays.

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Ok, so moving forward to Mixing / Editing.

Again this is hopefully read by the novice, and or people trying to increase their knowledge in the gear dept. and techniques.

So,

1. You have chosen a piece to record, and you have picked the winning sound, and what's this ?
My track, no matter where I set the volume slider, doesn't quite sit in the mix.

What can I do about it ?
Well, you add a compressor of course. Which makes the overall sound more even, or louder, or softer. Which ever way you need it.

What is missing, that doesn't quite make it sound STUDIO enough ?
Or, what is missing that doesn't make it sound LIVE enough ?
The answer depends on the actual circumstances for the question, but, most can benefit from effects (plug-ins). Of course it could be they way it was recorded.

2. My track is too loud no matter what.
What can I do about it ?
Again here would be a good use of a compressor, in limiting formation. Keeping a ceiling on them.

3. The major labels sound good in my car, but my tracks don't.
What am I missing.
Well, this is a lot to answer depending on the usage of equipment (CD,Ipone,RF broadcast, bluetooth, headphone-cable to radio/deck. ETC.
But if it is CD, I would say dithering, maybe EQ also, but, here is what dithering does. It makes use of how digital equipment (CD player) decodes the digital music, BACK to analog.
Because Digital music is a REPRESENTATION of the analog music, there can be problems.
One of those being, digital truncation distortion. A nasty sound in the lower end of the bandwidth.
Dithering 24 to 16 bit music, is a MUST. 32 floating point and up, not as much unless bouncing down to less. Like 32 to 16, for CD production.
I can't fully explain dithering to you, but, depending on your setup, you probably should be using it.

4. My tracks sound good in my car, and my home system, and my computer, but NOT my mp3 player.
What went wrong ?
Well could be the dithering again, but probably EQ. The reason is different speaker sizes warrant different EQ.
Mostly, the ol' earbuds don't represent the song like bigger speakers.

5. My wikiloops tracks don't sound good when others add in.
Well here is another forum post, but dithering could have helped if all used it.
Meaning, the aggregate total of all the non-dithered tracks adds up, during each overlay.
Or, it could be an inexperienced mix, where each one adding tracks, not mixing correctly adds up also.

Didn't mean to interrupt the existing thread, but, thought even though we have gear, are we getting the most out of it ?

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@ Danny thanks for sharing that !
The beloved RP-1, my friend had that when it first came out.. Wow, brings back memories.
Metalica days for me. Garage days, the black album.
Anyhow, the 11rack has grown in interest, and coupled with pro-tools I am sure sounded great.
But, which emulator do you use ?
I use GTR3 from Waves, it is the sound on most of my recordings on here. I have used the PODHD on a couple.
I really like the intuitive interface. Click on a pedal, turn the knobs like a real pedal, and save. Click on the Amp, turn the knobs like a real amp, and save.. You get it.
Have been contemplating on using it live, on a chromebook or like device.

For Bass I use a Hartke pre-amp pedal, the "BASS ATTACK" pedal. Sounds just like my Hartke rig.
I haven't released any bass here, but might in the future. So many great bassists here already. Like DannyK, for one ! :)

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Hey Martin,
Put one of your examples and explain what you think your 'isolated' sound is.
Would really help out this thread.
And maybe the collective can be positive to get rid of the isolation.

Again , I enjoy your drums, and think they sound good. But things can always be improved upon. And if you come out happier, then more of the better.

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I use Amplitube. I have the Ampeg collection. The rest is kind of you to say but I'm just a beginner - lots to learn!

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I'll answer more comprehensively when I'm not on my phone!

But initially, I do wonder whether I should add drums 'as is' both as HD and the mix MP3. As stems get added to and mixed and remixed, the kit gets more and more processed with each add. I do wonder whether supplying the drums 'as recorded', bar the basic compression, reverb and EQ I add is of benefit to others. I suppose that's what the HD track is for which, after Oli's recommendation, I supply with no reverb and only the standard compression, EQ and gates I use.

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@MasterK

I'm confused...

I did some reading up on dithering and it's slightly confusing as to when one should or shouldn't use it. It appear that Reaper does a lot of this 'behind the scenes' for you as you work, transparently doing the transcoding. But, unless I'm reading something wrong, should you only use/need dithering if changing the sample rate of the project or does it apply to bit rate too?

I record at 44.1khz/24-bit. My mix down renders are 44.1khz/16-bit for HD WAVS (for space reasons - many tunes a 24-bit wav won't upload as it's too big) and the MP3 mixes at max quality the LAME encoder supports (320kbps). So, during my chain of recording I should never leave the realm of 44.1khz, only moving down from 24 to 16 for mix down at time of render.

I suppose my other question is, do I need to identify any incoming mix MP3(s) (i.e. the templates I've downloaded) for their bit/sample rates and dither them for use in Reaper? From what I've read, Reaper takes care of this automatically and many recommend that Reaper's built-in (and automatic) dithering is more than good enough for this purpose.

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I'm back home now. I'm going to record a track later today then upload (not to the Loops) the recording during their various stages of mixing, from raw, to compressed, to reverbs to multi-band compressor. I'd be keen to get your opinions on what's right and what's wrong.

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When i play i normally just use one of the presets im used to. That can be factory or homemade or one from axechange. Since i normally play at the computer i have Axe-Edit open all the time so tweaking is very convenient and doesn't take much time once you are used to to Axe-Edit.

The situation where i spend time tweaking is when im experimenting with new sound ideas. Like replicating Scott Hendersons setup with separate cabinets for wet and dry signals.

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Hey Marty !

Yes it is all about the bit rate. Not resolution.
8-16-20 all need dithering. Only 24 bit or (32 bit floating) DO NOT, until time to hear it on a CD or mp3 player..
That's when we bounce it down, to HEAR it.
There may be a 24 bit D/A converter but it will be EXPENSIVE.
But to listen we need 16 bit, humans hear 16 bit the best, 24 bit has harmonics that are out of range for our ears.
I hear what your saying, but hang in there. (Think CD) not FLAC and loss-less compression.
(24 bit FLAC and the like, do add some music dynamics/sounds, but, not like they want you to believe.
Mp3 makes a decision based on an algorithm, cutting un-needed info.
This causes some sounds to be cut away for space)

This is the reason people will argue about 24 bit. Don't get me wrong, it is better to the most point for editing.
But is used so we don't have to dither, until lowering the bit rate.

That is the paradox, conundrum etc....

Think like this.... See if it helps.

44.1 Khz. is like listening to your music out of speakers with one ear, and plugging that ear, with your finger 44.1 thousand times per second.
That is what your D.A.W. hears during recording, or sample rate.
The Bit rate is how much INFO can be stored, PER sample.
24 bit quantizes the info, or lowest subtle audio, to the atomic level, so no dithering required.
The audio sine wave is CONSTANTLY changing. And the recording shows that, hence the sine wave we see. ~~~~~~~...
SO.
The differences in the most subtle sound, the one that doesn't quite make a blip, or 1 bit, is rounded up..
Which during playback makes square waves, because it's truncated differently, than that of the ORIGINAL sound. (Digital issue).
The one issue, that most audiophiles, will say that's why digital, should be thrown to the wolves..

So, to COUNTER that, we dither.

Add noise to up-the-ante on the low ones.

ADD NOISE you say.. HOW ABSURD... One would think.

BUT...

This makes for less SQUARE WAVE patterns at playback.
This is what the less than 1 bit recordings do. They make square waves.

That being said, any RECORDING, that is under 24 bit, must be dithered when you bounce down to 16 bit.
The one we use, to have human ears hear it from a D/A converter (CD player) (Digital I/O interface), mp3 player, our favorite toys on here.

YOUR input/output device probably plays back @ 16 bit. (I know some will argue) I will wait for it.. LOL..
But the human ear can only hear so much. NOT 32 Khz. (Ultrasonic).
People think they record in 24 bit, and hear it in 24 bit. Not usually true..

And that is backed up here...
https://xiph.org/~xiphmont/demo/neil-young.html

This kills the whole promotional, 24/192 thingy.. For the most part.

So , the higher the RESOLUTION (44.1 Khz.) or (48 Khz.) the MORE digital DISTORTION is recorded..
More earplugs per second, and more info recorded, that turns into errors.
And 24bit adds more harmonic distortions, and at 192Khz.
That's a HUGE FILE. none the less.

That is why we drop down, 24bit is great, and doesn't need dithered, until downgrading the bit rate, to export into lower bit rate environments.

I hope that helped.. And didn't muddy the water.

Look to www.earlevel.com to help with all the details.. They are the experts !

That is where I learned about it finally. When it clicked.
And there is a cool tool, to animate and let you see why.. Called the Dither Widget..

And yes I would enjoy helping you figure out what is going on with your tracks.
How about you create them on here under INVITE only..

But, here is how I would approach the experiment.
Take a DB meter, or get a free app, and measure from your speakers, to keep ALL output at the same 85db, slow response, weight c, for human ear simulation.

This way it's not an AMPLITUDE judgment. As these TEND to go..
ex... I recorded this EXTRA hot and it sounds better... etc. etc..
Not judging you or anything.. Just, making a even playing field to judge, so to speak.

And now, to the end, of the dithering question....

During mix-down, you should dither every track, and the MASTER..POST-FADER..
Did I say that out loud... POST-FADER ....

This makes the tracks have the LEAST amount of SQUARE waves going on..
So when you import the other .mp3s, and add up the distortion, it becomes aggregate, that is what's going on when we hear artifacts. etc.

There should be a 101 on this.. Oh wait, that is what we are all doing here.. :)

I appreciate the open discussion we are all having.

Yes, I could be wrong about some of this, it is my understanding, and I am not getting paid to make tracks, so at some point, I quit learning about dithering, and just use it. :) (my disclaimer for liability.) :)

One more thing, your stems, or stereo mix-down, should be free of ALL effects, again to the highest loudness (-0.3dbs) with dithering, and multi-comp, POST fader.

This is NOT to be confused with track NORMALIZING.

Normalizing is loudness, with a certain Db level, to create the Signal/Noise ratio.

You should dither AFTER normalization.

Dithering is removal of DISTORTION.
Distortion is QUANTIZATION ERRORS.
Distortion is square waves. Bad news.

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Cool Nilton,
Thanks for sharing that.
It seems as if, when one is comfortable, or over the initial learning curves, or has great options at hand, they feel as if it is easier. I do.

Even after the next piece of gear comes along, we get acquainted with it, we become content, and play more.. :)

And that brings me to the next facet.

Mix-down / Editing.

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Now that we have our content, what do we do ?

We edit, and mix-down to stereo !

Then we convert to the format we want. mp3, FLAC, etc.

That is the short answer.

The real answer is there is a lot between A and B.
And there is an answer about this, 3 times the number of musicians/ producers /masters.

But the stages are what's important.

Technique is what the discussions are made of.

1. Edit each track, or stem, until you are happiest with the results.
Adding a compressor, or effect, or other plug-in type of editing / processing tool.
Then normalize, then dither. In that order.

My technique is this, I try my best, to make it as loud as possible. To max out the meter. To (-0.3 db's on the track, after all effects and or processing have occurred.
Others do leave some dynamics headroom. Which can introduce our friend distortion, because it lowers the most subtle representation, to under 1 bit. This causes the math issue, or , digital distortion truncation.

2. Add other tracks, or stems, until you have enough content.

Is there too much, well, yes. Depending on your setup, processing power, D.A.W. etc.

3. Mix each together, until the result is pleasing, without errors (clipping) on the master stereo out.
Let your friends hear it, and give feedback.
Upload to www.Wikiloops.com, and get a lot of feedback, from like minded people.

My technique here is, to watch the master out meter, to make sure the "Mix" is not going above -0.3 to ensure no clipping occurs. If so, find the culprit by muting.
This doesn't change anything, and helps narrow the field.
When you do find the one, or more, check for or add a compressor or multi-comp.

4. Master it, this is not MIXING !

This is setting it up, for a certain export. As discussed earlier.
And is the SCOPE OF WORK, so to speak.
That can change. Due to the nature of EXPORT target audience.
Again, add processor plug-ins to make it yours.
Listen to major labels of the same genre, and copy-cat them.

I utilize a bunch of different plug-ins, and you can never explore them all, too many.
Best practice is use as little as possible, because every edit here, and processing adds MATH !
And.....
You guessed it, this adds the dreaded distortion artifacts.
The algorithm doesn't know you put a whisper track in for a special effect.
It only knows MATH ! 1's and 0's .. That is all.

So, unless your just going to hand it to an engineer at 24 bit, you must dither !

Utilize at least a compressor (PRE fader), and multi-band compressor/limiter (POST fader).
And at least a Parametric EQ, and/or a 10 band graphic EQ.
Sometimes I use a 30 band EQ, more adjustments, but more work/time.
These are your friends in the toolbox. Find ones you can deal with and use them.
Explore others when you just get so tired of them.
Then add in some effects on final, like stereo, surround, and the like.
Make sure you add a dithering tool (POST fader , after the multi-comp.)

5. Distribute the results.
We all want others to hear it ! :)
And we all want to hear others work, it helps us with ideas, and self-adjustment.

Of course this is up to the master engineer, YOU !

Share your process and/or questions !

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Yes, let's do that !

Read the above post, I have some agenda's in mind on how !

Let's create at a certain DB reading 85.. and keep the amplitude steady. :)

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Yes Sir.
That is why we all do this, to learn the next piece of the pie.

Your sounding good at this end.
I like the sounds you choose, and they sound good enough that I couldn't tell, if your recordings are amps, with mic, or direct to desk, with effects.
You have an ear for setting it..
Maybe you could share your approach to getting the sounds you use, or initial setup of a track.

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And I don't even have a subwoofer, lol. I guess what I do that some may not is use a reference track. I try to match my levels with a a professionally mastered song of similar style. That should help with levels, at least. Martin has told me I blend in well, too.

I run both of my basses through a SansAmp Bass Driver/DI box. All dials are at 12 O'clock except the blend which is at 3 O'clock (75% wet). That goes directly into my audio interface. I have two basses and they are night & day tone-wise. My P-Bass has flat wound strings on it. They really punch through the mix and have short sustain. With a foam mute it can almost give an upright type-of-sound. (I sometimes use the mute a la Paul McCartney, Carol Kaye, etc.) The secret for that bass is all in the strings and fingers. I don't use any effects on it whatsoever.

My other bass has "regular" strings; nickel roundwounds. It has the more modern rock sound and, with a plugin, metal sounds too. I often use a pick with it so the dynamics are a bit different - it needs a bit of compression (it's difficult to strike the strings with a plectrum with the same pressure each stroke when using alternating picking). For that compression I use a plugin from EZMix. I have the threshold dial at 1 O'clock and the ratio at 12 O'clock. I don't know what that would be but it's probably around 4:1 if I had to guess. For the metal sound/growl I also use a distortion plugin from EZMix that was designed by [url=https://en.wikipedia.org/wiki/Randy_Staub]Randy Staub[/url] himself.

Then, when it comes to levels, if the original remix is within normal limits I set my levels to not go above -12. That's about where I achieve unity gain between it all. Oh, I pan my bass -7, also. This helps throw the ears off a bit, I believe, and helps the bass stand out without it being right in front of your face.

I hope some of that makes sense. I work in Logic Pro.

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@ DannyK,

Yes that makes sense, and thanks for explaining.
Everyone's approach is different, for a number of reasons.
And if it helps one person get over some hurdles, then we have done good.

What people have to keep in mind is every instrument is different, and the sound coming from it is unique.
Then add a I/O device, and tweak it to make recordings sound good, takes some time.
So if someone has a similar device your settings may help them get there..

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