#1
I may be making more out of this than necessary but it really bothers me when I record a song and play it back and I'm like sweet it sounds good and then I play a song on Itunes or whatever else and its like twice as loud. I've tried various things and usually I just end up making it sound worse with clipping etc.

A little background info, I'm using Hydrogen for drums. Using Audacity and playing through a mic'd tube amp. I have yet to add bass to any of my recordings...currently saving for a bass guitar and amp. I realize adding bass would help but I don't think its the main problem.

Thoughts?
#2
A large part of that is mastering. Mastering is the "finalising" process of mixing, one part of which involves boosting the volume and limiting it to the desired level.

Bass is absolutely going to be a large part of why you aren't getting close. Depending on your interface you may not need a bass amp, a lot of guys these days just plug direct into the interface and use an amp sim. Bass frequencies take up a large amount of headroom, so even if you were to get one of your tracks professionally mastered it would still sound quiet in comparison to commercial mixes.

Assuming you actually want to learn how to master, rather than simply increase the volume of your mixes, I would suggest downloading the demo for Izotope Ozone. It's an all-in-one mastering plugin that many pros use. Not very expensive at all for what it is. Plenty of tutorials on Youtube as well.

Again, mastering is the FINAL process of mixing. Given that you haven't actually recorded any bass (let alone mixed it) I personally wouldn't bother looking at mastering until you get the other two processes down sufficiently.
#3
ryane24 Firstly, you need to maximize the input signal for whatever sound source you are recording. So if you are recording your amp through the mic, your interface's input gain for the mic should be set just below clipping upon the hardest strum of the guitar. That will give you as much input signal as possible to work with in the mastering stage. Recording soft parts and loud parts separately will help you as you can get more input gain on the soft parts.

Secondly, after you are done mixing the tracks (compression, eq, effects, etc.) and everything is set in volume properly relative to the other sound sources, you want to normalize your track. That will set the highest volume peak of the mastered track to a dB level of whatever you set it to. Typically that is -1dB, right below clipping of the software audio drivers, which clipping happens at any level above 0dB.

If you have really large volume spikes but the rest of the track is of a relatively low volume, you may want to use a bit of compression on the master track as well. With the large volume spikes, the normalization will cater to those spikes and really the master track volume may not increase that much. Using the compression will bring down the large volume spikes closer to the softer volume of the rest of the track, allowing the normalization to have more of an effect.

And yes bass is absolutely essential for most genres of music.
Last edited by Will Lane at Jan 7, 2017,
#5
I appreciate the great advice guys.
Random3 Depending on your interface you may not need a bass amp, a lot of guys these days just plug direct into the interface and use an amp sim.

When you say amp sim are we talking about a peddle? If I could avoid buying a bass amp that would be sweet since I only plan to use it for recording.
#6
Quote by ryane24
I appreciate the great advice guys.

When you say amp sim are we talking about a peddle? If I could avoid buying a bass amp that would be sweet since I only plan to use it for recording.


No, an amp sim is a plugin or piece of software that simulates the signal processing of a traditional amplifier. Just realised you use Audacity, in which case this isn't an option.
#11
Hmmm.... some misinformation here,

First, I would be surprised if any pro mastering engineers used ANY all-in-one so-called "mastering" package. They are aimed at bedroom producers who just want a quick dial-in and don't actually have the experience and know-how and equipment to get it there any other way. The real pros will use dedicated mastering rooms with dedicated mastering gear.

Second, the signal you record should NOT be as hot as you can get it before it clips. This USED to be true in the days when people recorded to tape, because tape is, by it's very nature, hissy. Getting as loud a signal to tape as possible achieved two things - it made for better signal to noise ratio, and based on the way tape behaves, added some natural compression to the process.

It also USED to be true in the days of 16-bit digital recording. Again, this had to do with the distance from the noise floor to the maximum amount of headroom. With 24-bit recording, the distance is exponentially greater, and you do not need to get it anywhere NEAR that hot. Add to that, the fact that analog gear (ie. your preamps and stuff.... anything that happens before the signal hits your digital converters) is rated to work optimally at a particular level. That level is +0db. After that, it begins to distort. 0db in analog is NOT the same as it is in digital. If you ran a true 0db sine wave through an analog channel and recorded it to a digital device, the top of your signal would actually be somewhere around -12db on your digital meter. In other words, the BEST level for you to record at in the digital domain is one that places most of your program material between about -15 and -20db on your digital meters, with the peaks not really going much beyond -12db.

When you run your analog equipment past that level, it is working harder than it is designed to, and the consequence is distortion, etc.

To the original question, the answer is compression. Now, mastering is considered a whole unique process that happens after mixing - not the last stage of mixing - and involves more than just compressing, but this is what we're looking at.

If your goal is to get your recordings louder, place a compressor (or two... or three... ) on the master bus followed by a limiter on that same master bus. To make sure you don't take all the punch out of your mix, be careful about setting your attack times too fast. Somewhere around 20ms is usually a good place to start. Two or three compressors at moderate settings will be a lot more transparent than one compressor that is simply beating the bejesus out of your song.

To elaborate a little, mastering is the process of:
- deciding on the best order to put songs in
- deciding how much of a gap to place between each song
- adding ISRC codes etc that identify the disc to retailers, media, etc.
- adjusting edits and fade-outs at the beginnings and ends of each song ("tops and tails.")
- adjusting EQ so that each song sounds like it belongs together as a consistent and cohesive whole
- compressing tracks to adjust levels for playback relative to each other
- copressing and limiting tracks to make them loud enough for "commercial release."
- creating a DDP file (like an audio version of an ISO) that will be used by the manufacturer to make the discs.

CT
Could I get some more talent in the monitors, please?

I know it sounds crazy, but try to learn to inhale your voice. www.thebelcantotechnique.com

Chris is the king of relating music things to other objects in real life.
#12
Adding to excellent explanation from axemanchris above - as long as you have no bass in your recording you need to compress the other tracks a lot. But remember after you will put the bass in - which brings a lot of dynamic energy itself - you will have your master bus overdriven for sure so you will have to reduce amount of compression accordingly.
#13
Quote by axemanchris
First, I would be surprised if any pro mastering engineers used ANY all-in-one so-called "mastering" package. They are aimed at bedroom producers who just want a quick dial-in and don't actually have the experience and know-how and equipment to get it there any other way. The real pros will use dedicated mastering rooms with dedicated mastering gear.


I can name multiple mix/master engineers who do all their work entirely in the box with plugins such as Ozone. Apart from that I agree with your post.
#14
Quote by axemanchris
It also USED to be true in the days of 16-bit digital recording. Again, this had to do with the distance from the noise floor to the maximum amount of headroom. With 24-bit recording, the distance is exponentially greater, and you do not need to get it anywhere NEAR that hot. Add to that, the fact that analog gear (ie. your preamps and stuff.... anything that happens before the signal hits your digital converters) is rated to work optimally at a particular level. That level is +0db. After that, it begins to distort. 0db in analog is NOT the same as it is in digital. If you ran a true 0db sine wave through an analog channel and recorded it to a digital device, the top of your signal would actually be somewhere around -12db on your digital meter. In other words, the BEST level for you to record at in the digital domain is one that places most of your program material between about -15 and -20db on your digital meters, with the peaks not really going much beyond -12db.

When you run your analog equipment past that level, it is working harder than it is designed to, and the consequence is distortion, etc.
I am not sure I am understanding you. You say that analog inputs work best at 0db. Are the input gain controls on an interface like the Mackie Onyx Blackjack or a Focusrite Scarlett controlling an analog input or a digital input? If analog, I do not see the problem in what I said compared to what you said. The analog input is set as high as it can go (0db) before it starts clipping. If they are digital inputs, then I am confused. I would think that the clipping/overload light would light up when the signal is clipping, does not matter which type of signal it is adjusting. Keeping the signal "green" no matter if it is analog input or digital input means there is no clipping happening to the signal. Correct?

Or maybe you are talking about the translation of the interface's signal into the DAW. If I normalize my track to -1db in the DAW, does that translate to -1db digital, but +11db (clipping) analog? I normalize my tracks like that all the time and I do not notice any clipping, especially considering the degree that the signal would clip at 11db over.
#15
Quote by Random3
A large part of that is mastering. Mastering is the "finalising" process of mixing, one part of which involves boosting the volume and limiting it to the desired level.



Okay, I gotta clear this up, because it's bugging me. Mastering is not "part of mixing". It takes place after mixing, ideally by a separate person.


Anyway, OP:

You need bass guitar. Your mix is gonna sound pretty empty without it. It won't make it louder, but it'll make it better.

And can I recommend you stop using audacity? Reaper is extremely cheap and is much more full-fledged than Audacity.
Current Gear:
LTD MH-400 with Gotoh GE1996T (EMG 85/60)
PRS SE Custom 24 (Suhr SSH+/SSV)
Ibanez RG3120 Prestige (Dimarzio Titans)
Squier Vintage Modified 70s Jazz V
Audient iD22 interface
Peavey Revalver 4, UAD Friedman BE100/DS40
Adam S3A monitors
Quote by Anonden
You CAN play anything with anything....but some guitars sound right for some things, and not for others. Single coils sound retarded for metal, though those who are apeshit about harpsichord probably beg to differ.
#16
Quote by Random3
I can name multiple mix/master engineers who do all their work entirely in the box with plugins such as Ozone.


I'm all ears. Not saying your wrong at all. Just fascinated to see who these Ozone "pros" are and what they've done. I did a quick google search for Bob Clearmountain and ozone, and Bob Ludwig and ozone and came up dry.

CT
Could I get some more talent in the monitors, please?

I know it sounds crazy, but try to learn to inhale your voice. www.thebelcantotechnique.com

Chris is the king of relating music things to other objects in real life.
#17
Will Lane

Okay... we may be talking about meters in two different places. If you are looking at your meters on your interface or on a mixer, then you are in the analog domain. You're right in that you don't want too low a signal there.

I was talking about the meters in your DAW - which I think is where most people do their metering. If you're hitting peaks in your DAW upwards to -5db, I can pretty much you're clipping something somewhere.

Now, you can still muck things up without clipping the inputs of a mixer or interface. What some people do is adjust the input using the trim control and meter on the interface (the one that measures input signal), and then using a master gain or a channel fader to boost the signal even further. You'll get upwards to 0db on your DAW meter, no clipping on the front end of your interface, but will still be driving your gear beyond its optimum range.

CT
Could I get some more talent in the monitors, please?

I know it sounds crazy, but try to learn to inhale your voice. www.thebelcantotechnique.com

Chris is the king of relating music things to other objects in real life.
#18
Quote by axemanchris
Will Lane

Okay... we may be talking about meters in two different places. If you are looking at your meters on your interface or on a mixer, then you are in the analog domain. You're right in that you don't want too low a signal there.

I was talking about the meters in your DAW - which I think is where most people do their metering. If you're hitting peaks in your DAW upwards to -5db, I can pretty much you're clipping something somewhere.

Now, you can still muck things up without clipping the inputs of a mixer or interface. What some people do is adjust the input using the trim control and meter on the interface (the one that measures input signal), and then using a master gain or a channel fader to boost the signal even further. You'll get upwards to 0db on your DAW meter, no clipping on the front end of your interface, but will still be driving your gear beyond its optimum range.

CT
I think we are thinking similarly know. However I am still a bit confused and skeptical as to what you are saying- not trying to downplay your experience but maybe I am just not understanding you correctly or some other facet of this. You are saying that if I set the master track output to ~0db in my DAW, that it will clip the analog inputs of the equipment afterwards? As in, 0db on a digital meter when translated to analog will be +12db which would be clipping the inputs?

Do digital/analog conversion drivers not automatically bring down the signal level to compensate for the differences between digital max gain and analog max gain? Further more, I was under the impression that adding db above 0 for analog is not necessarily clipping the inputs.
Last edited by Will Lane at Jan 11, 2017,
#19
It took me a while to find this, but I knew there was a really well written article on this. Note that the author is very credible and the article itself had gotten picked up by ProSoundWeb, which is also a highly credible source.

http://www.massivemastering.com/blog/index_files/Proper_Audio_Recording_Levels.php

CT
Could I get some more talent in the monitors, please?

I know it sounds crazy, but try to learn to inhale your voice. www.thebelcantotechnique.com

Chris is the king of relating music things to other objects in real life.
#20
Quote by axemanchris
I'm all ears. Not saying your wrong at all. Just fascinated to see who these Ozone "pros" are and what they've done. I did a quick google search for Bob Clearmountain and ozone, and Bob Ludwig and ozone and came up dry.

CT


Joey Sturgis and Skrillex are the two that come immediately to mind. For plenty of guys who do mastering ITB, Ozone is pretty damn convenient.
#21
I think until you have the bass recorded all the talk about mastering is moot. Bass and kick drum occupy so much of the overall sound that just trying to master a half finished song will not seem very loud. I do a lot of mixes as I go along on a project and make a CD so I can listen in my car or on a normal audio system (not my studio monitors) but volume is not one of the parameters I am looking at until it is final or close to final mix. Also don't get trapped by the "loudness" issue. Look up "loudness wars" on YouTube to see what I am referring to. Some of the loudest CD and MP3 tracks get that way by being crushed and processed so heavily they lose a lot of dynamics. This is happening on some of the biggest albums being released today. Don't worry about the volume of the overall project until you are nearing completion. Just record your individual tracks at a decent level without clipping. Personally I use T-Racks Mastering Suite for final mastering but I only use the individual VSTs in the package not the automated presets. Nothing wrong with the presets but I prefer to set my own levels on EQ, compression and the final Brickwall compressor myself depending on the song.

I think Audacity is great for starters to get you going but you do outgrow it quickly and I find the included VSTs often confusing but very workable with patience..
Yes I am guitarded also, nice to meet you.
#23
Quote by Random3
Joey Sturgis and Skrillex are the two that come immediately to mind. For plenty of guys who do mastering ITB, Ozone is pretty damn convenient.


heh... cool. Now Sturgis uses it in conjunction with a bunch of other stuff, and for the most part, his discography is mostly pretty minor stuff - relative to the Adeles, Metallicas and U2's of the world. Respectable nonetheless, though. Skrillex has often been a guilty pleasure of mine, but fan or not, his stuff sounds fantastic, and you can't argue with a collection of six grarmmy awards. And it sounds like he uses it almost exclusively. Good on him.

CT
Could I get some more talent in the monitors, please?

I know it sounds crazy, but try to learn to inhale your voice. www.thebelcantotechnique.com

Chris is the king of relating music things to other objects in real life.
Last edited by axemanchris at Jan 12, 2017,
#24
I really appreciate all the advice guys. Since my original post I have started using reaper and done some research on basic EQ. I have also purchased a bass guitar and used a free amp sim to add bass to the recording. However, I think I'm going to end up getting a bass amp and recording via mic. The amp sim has left something to be desired, mostly because of latency issues.

Since downloading reaper I feel like a cave man who just discovered fire. It's a little overwhelming, I've found myself obsessively adjusting things until I don't even remember what it sounded like before! Can't believe I've been using audacity this whole time.
#25
ryane24

Recording via amp/mic is not going to fix the latency problem. If there is a latency issue, the computer doesn't care whether you have a bass guitar plugged in or a microphone plugged in.

I would be able to tell you how to fix this if it were Pro Tools, with Reaper however you will need to wait for someone else I think!

Unless there is are options in Reaper for Low Latency Monitoring, Delay Compensation and Buffer Size I wouldn't know where to go.
#26
It might help with the latency issue, actually. Consider that the amount of latency is proportional to the amount of "stuff" the computer is asked to do between the time the sound enters the computer and is then spit out of the computer.

If it just needs to go in, get converted to digital, read and stored to the hard disk and sent to speakers, that's pretty easy. Even low-latency monitoring with 10ms of delay should be pretty well fine for most people.

However, if it needs to do all that, AND get processed through an amp sim, that might increase your latency to 30 ms, which is really getting into *really* pesky territory.

CT
Could I get some more talent in the monitors, please?

I know it sounds crazy, but try to learn to inhale your voice. www.thebelcantotechnique.com

Chris is the king of relating music things to other objects in real life.
#27
Yeah, I record all my guitar tracks with an amp and a mic and I haven't had any issues with latency. But I'm also not really doing any real monitoring. I listen to the track via headphones and my amp is loud enough to where I can hear it outside the headphones.

The issue with the amp sim was with monitoring but also the recording was like significantly out of time to the point where I would record without the monitoring and then snap the track to fit the other tracks.
#28
The simple answer, buried above, is that you need to add a compressor and a limiter on your master buss - that alone will drastically increase the perceived volume. Also, EQ out lower muddy frequencies since those will cause your track to clip much faster.
#29
ryane24 does your interface have direct monitoring ( i.e its own mixer software)? If so simply use that when tracking bass, don't engage the input monitoring on the Reaper bass track - that way there will be no latency at all.