Mixing Hacks Roundup #1 - EQ

I love "best of the best" roundups.

What I'm about to show you here is a round up of the best videos I've come across for how to use EQ in mixing. I'll come back with another one for mastering.

First - here's a great hack called "The Sweep". The trick to this hack is to listen for what you don't like, find it by boosting until you hear what you didn't like clearly, then do a cut. In this video, Joe cuts some thump out of an acoustic guitar. The reason this is what he doesn't like is that it will interfere with the kick drum and bass guitar in the mix. There are other far more offensive frequencies that he boosts, but he explains why he leaves them alone. If It were me, I would have treated one of them. But it's not me, so ... enjoy!

And you've got to love the internet. Here's a different take on the Sweep. This is more of a mastering level video, but the principal is transferable to individual tracks.

And here is a couple of Hacks in action.

Now I will say that before you even get to EQ, you should have a rough balance done of your mix. Do this in MONO. Then use subtractive EQ on each track in the context of the other tracks to fix problems like muddiness, or notch our honkiness and squeakiness etc.

What is Subtractive EQ? Glad you asked

Something you need to know about is Resonant Frequencies. Removing those will really clean up your mixes!

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Home Recording Production Advice

Home recording can be challenging. As a DIY musician, I understand! So, I've compiled a page of some of the best free advice out there for getting the most out of your home recording gear.

Take a look!

http://www.audio-mastering.net/p/blog-page.html

Audio Production Podcasts: My Top 4 Favourite

I can't count the number of times I've seen lists of 20 or 50 *recommended* podcasts for listening.

Who has time for that?

As an educator myself, I hate unnecessary fluff.  Large lists of podcasts are just that. It's impossible to keep up with more than 4 or 5 podcasts at a time, unless you are a long distance runner, drive a semi for a living, or have nothing better to do than listen for hours on end to podcasts in your spare time.

Another factor I go for is brevity. If it takes you two hours to say what could have taken 20 minutes, I'm out. Keep it simple and keep it short. I think that'll be my new motto.

I've whittled the list down to my top 4. To make it on this list, the podcast needs to be educational in nature and generally interesting to listen to.

1. The Mastering Show - http://themasteringshow.com/
I love, love, LOVE this podcast! Ian Sheppard is a mastering engineer and a natural educator. He is able to distill complicated concepts down to simple anecdotes that are not only easy to understand but also highly actionable. This is almost a master class in Mastering. If you are into mastering, or just want it demystified, I can't recommend this podcast enough! Ian has several products that I highly recommend as well.

2. Recording Revolution's Youtube Channel - https://www.youtube.com/user/recordingrevolution
This guy is prolific. And an excellent educator as well. Highly, highly recommended. Grahm's vlogs and products completely changed how I approached recording and mixing, for the better!

3. Recording Studio Rock Stars - http://recordingstudiorockstars.com
I'm only a couple of episodes into this interview style podcast produced by Lij Shaw, but I've found the information already to be invaluable.

4. CD Baby Podcast - http://cdbabypodcast.com/
This podcast is a bit long winded, but they deal with a lot of music industry business stuff, which is important. CD baby is an important part of my own journey as an independent artist.

There you have it, my 4 podcasts faves!
enjoy!

Metering Plugin (and Free Mastering Offer!)

There is a new metering standard called LUFS, which is quickly becoming a standard loudness measurement unit for mastering. Here is everything you never wanted to know about LUFS: https://tech.ebu.ch/loudness

Hoffa 4U makes a nice LUFS meter that packs a bunch more features, including some Mid/Side processing and panning.  It's called HOFA 4U Meter, Fader & MS-Pan.


https://hofa-plugins.de/en/plugins/4u/

Feeding a good limiter so that you get just enough gain reduction where it's not distorting and a reading of about 10-13 LUFS on the 4U meter will make any rock, pop or EDM song plenty, plenty loud while retaining clarity and punch.

I use this along with the TT Dynamic Range Meter to measure the peak-to-loudness ratio of the music I am mixing and mastering. It's well worth the price (free!).

If you are interested in having me master a song for you - for free - head over to www.levityproject.com/mastering and fill in the mastering form.

Here's to your music!
Ryan

Dynamic Range (and Bit Depth Part 2)

Ah the boring technical aspects of audio production. I love it!

So before I talk about Bit Depth, I want to talk briefly about dynamic range.  Dynamic range is essentially the difference in volume from the loudest to quietest portion of audio in a waveform at any given moment. This is also call this the crest factor.

When people talk about the dynamics of a song, they tend to be talking about how loud or squashed sounding a song is. Songs with wide dynamics tend to sound open and airy. Songs with limited dynamics sound punchy and hyped in the best case to squished and crunchy in the worst of cases.

An example of hyped and punchy is Pearl Jam's Dissident. It has a dynamic rage of 8, which is about the minimum level of dynamic rage. Higher numbers are even better in this day and age of volume normalization and web streaming.

An example of squished and crunchy  is none other than Metallica's Death Magnetic. If you look it up on youtube, fans remastered the album from guitar hero stems and it sounds way better than the PRO master! The below video shows the difference between the retail version and the less squished guitar hero version. The difference is stark!!

It pains me so see a lot of Christian recording artists and Worship artists becoming victims of the loudness wars. At work, we sometimes put on Praise 101 and I can stand it for about 30 seconds. Everything is hyper squashed and distorted. Ugh!

All recordings have a dynamic range before mastering, and it's the job of the mastering engineer to decide if the dynamic range is well suited for the song and the album, or if the dynamic range needs to be increased (made quieter) or reduced (made louder). Mastering Engineers can do both. Reducing the dynamic range is quite a bit more difficult than increasing it, FYI.

So when someone talks about "squashing the dynamic range" or "over limiting" or "over compression", what does it mean?What we are really talking about is taking the dynamic range that a track already has and reducing it too much. As a rule of thumb, around 8db-10db of dynamic range is plenty good.

Here's to your music!
Ryan

Bit Depth (And Sample Rate Part 2)

This is part 2 of my previous post on sample rate. 

 The video below shows a full 16-bit Game sound sample. The difference from the previous samples is insane. You have vocals, guitars, drum kits. But notice ... it's just a little bit ... washy sounding.  

OK that was a trick ... if you have the quality set to 480p, the washiness should go away, mostly. But now that you are aware of it, listening to youtube music on low fi settings will now dive you mad. Sorry!

OK still with me? Here's the thing. Bit depth makes a huge difference. And 32-bit is even better than 24-bit ... 4 billion values of depth is nothing short of overkill, but it has it's advantages (more on that some other time).

But what about sample rate? I talked about it in my previous post, but I didn't get into the nitty gritty other than to show you that recording at 192khz can be redundant. (although up-sampling during mixing and mastering can be helpful).

Sample rate is essentially the number of times a sound is captured by the sound card each second (bit-depth is a combination of the lowest Digital loudness value that can be written, and the sheer number of depth values that can be written). It stands to reason that at 44.1khz (which is 44,100 hertz), a bit depth of 16-bit is fine since you can write a little over 20 thousand more values than the samples available. 44,100 samples available, 65,000 levels of depth available. The math works.There is a lot of buzz about recording at 24-bit, and I do record, mix and master at 24-bit. But I mostly do it for posterity and for the fact that even small volume changes can introduce many more bits to the audio signal. The cold hard fact is that I have fans, guitar amps and room ambiance in my recordings which create a noise floor that is above 16-bit.

But I've been using 24-bit for so long that I'm going to just keep doing it. Besides, 2-bit is the new standard for HQ audio, so if even for that I will continue to use it.  It doesn't seem to take up that much more processing power ayhow. On the other hand, upping the sample rate makes every plugin work harder because it has to process millions of more values every second.

Make sense? Hope so, cause I'm moving on!

The more times a sound card (Audio interface I mean ...) captures a sound, the more crisp  the sound will be to the human ear, to a point. Sample rate (just like human hearing) It's measured in something called Kilohertz (abbreviated KHz). A CD has a sample rate of 44.1 Khz, which is a little over twice the 'sampling rate' of the human ear, which tops out at 20 KHz. 

Now the first bizarre thing about digital sample-rate is that the very top frequency produced will be roughly half the value of the sample rate. So a 44.1khz sample rate has an absolute top value of around 22khz, which means you can do EQ adjustments up to 22khz but no more using the 44.1khz sample rate. A 48khz sample rate lets you do adjustments all the way up to 24Khz - and that's obviously not really necessary.

BUT

The second bizarre thing about digital sample rate is a thing called "aliasing".

Here is an awesome video that shows exactly how aliasing works. See my last post for a test to see whether your sound card/audio interface deals properly with aliasing and inter-modulation distortion.

So where the rubber meets the road on sample rates is that while a 44.1khz sample rate is good, if the anti-aliasing filter used by your RECORDING INTERFACE is poorly designed, you will hear this nasty aliasing in your recorded music! Ouch!

Plus, it will screw up your monitoring!! It would be all in all a very unfortunate thing. But it's a thing that I experienced with two pieces of gear. Not to knock M-audio (I did email them many times and got no response ... so hey, you get what's coming). The Firewire 18/14 and the Fast Track pro. When I recorded at 88.2khz, the sounds would be crisp and detailed. But when I recorded with the same microphones, the same instruments, the same settings and a minute later at 44.1khz, the aliasing would be unbearable (and shows up all over the place in my first album!). Since I had low computing power, my only solution was to record at 88.2khz, then down-sample the material myself. I did that with two songs that didn't appear on my first album, but that are slated for release on my second album. I used the original parts, so I'll be sure to load them up and do a breakdown once it's released!

So the lesson here is simply this: record at 44.1khz only if you are sure that your gear has a good anti-aliasing filter and no inter-modulation distortion at lower sample rates. I would like to assume that most modern gear deals well with these issues, but you never know! If you can afford the extra processing, make 48khz your go-to for recording, mixing and mastering. After all, 48khz is standard for DVD, and mastered for Itunes recommends sample rates of at least 44.1khz, but higher seems to be preferred. Also, if you release on vinyl, it's good to have masters that are 48khz. 

I would also do a little research and use audio interfaces known for their quality. You don't have to break the bank, either. I love my tiny but powerful Focusrite Scarlett 2i2. If you need the bells and whistles, M-audio makes sturdy gear, but you might have to play with it to make sure you aren't introducing nasty artifacts into your recordings if you work at sample rates below 88.2khz.

Here's to your music!

Ryan

Sample Rate

At the precipice of the home recording revolution, there was one sample rate: 44.1khz. And there was essentially one bit depth: 16-bit. Before that, there was 8-bit, and very low sample rates (less than 44.1khz). Now days, audio semi-pros and home studio folks are being pushed the line to record at the highest possible sample rates. 

Before you reach for your settings and crank up your sample rate to 192khz,There are some things you should know about sample rate

This website gives a good overview of sampling theory. http://people.xiph.org/~xiphmont/demo/neil-young.html

 

Pay special attention to the inter-modulation section of the website:

 

If you're curious about the performance of your own system, the following samples contain a 30kHz and a 33kHz tone in a 24/96 WAV file, a longer version in a FLAC, some tri-tone warbles, and a normal song clip shifted up by 24kHz so that it's entirely in the ultrasonic range from 24kHz to 46kHz:

• Intermod Tests:
  
  • 30kHz tone + 33kHz tone (24 bit / 96kHz) [5 second WAV] [30 second FLAC]
  • 26kHz - 48kHz warbling tones (24 bit / 96kHz) [10 second WAV]
  • 26kHz - 96kHz warbling tones (24 bit / 192kHz) [10 second WAV]
  • Song clip shifted up by 24kHz (24 bit / 96kHz WAV) [10 second WAV]
        (original version of above clip) (16 bit / 44.1kHz WAV)

 Assuming your system is actually capable of full 96kHz playback [6], the above files should be completely silent with no audible noises, tones, whistles, clicks, or other sounds. If you hear anything, your system has a nonlinearity causing audible intermodulation of the ultrasonics. Be careful when increasing volume; running into digital or analog clipping, even soft clipping, will suddenly cause loud intermodulation tones.

Source: http://people.xiph.org/~xiphmont/demo/neil-young.html

In my own system, 44.1khz, 48khz and 96khz have no inter-modulation artifacts. This means that I can realistically work at 48khz or even 44.1khz and still have just as good clarity and quality as 96khz, which is great because it means I can worry less about CPU load!

 

Here's a video of 8-bit music. One of the problems with early digital audio was that it utilized poor down-sampling filters, and probably wasn't dithered, which is why it sounds so washy and awful at times.

 

Here is 8-bit with high sample rates (at least 44.1khz). The difference is pretty stark.

Here's an interesting video that explains in more detail how that Low-fi music worked.

Do Recording Levels Matter? Yes!

I could almost leave this post as-is, but there are too many questions left unanswered.

The first one is: what about 16-bit? Don't I need to push my levels way up in order to minimize noise? 

Answer: Yes, unfortunately. But try to be reasonable. Boost the levels until there is no clipping but the signal is healthy enough that you can't hear low level noise. Compressing on the input before the recording interface can really help with this. If you know how to use a compressor and are still forced to record at 16-bit, I would say it's a necessity to compress the signal before committing it to the hard disk. If you record at 24-bit, it's not nearly as much of an issue.

This stuff can be really dry and boring for people, but I think important to understand what's happening behind the scenes so that you know why you need to make certain decisions later on.

Recording level means different things depending on what kind of recording we are talking about.

Analogue recording uses a different scale to measure peak level than digital recording. Confused? Don't worry.

In analogue gear, the level of signal can be measured with something called a VU meter like this one ...  (if you remember from a previous post, audio signal going through cables is just very low voltage electrical impulses) ...

With the VU Meter above, going above 0dB will not produce that hard noise, because 0dB is not a hard limit in analogue gear (in fact, it's the sweet spot). "Wait, wait that makes no sense!" you say. The key to unlocking the mystery is that the digital DB scale and the analogue dB scales are completely different scales! The digital scale is annotated dBFS, while the analogue scale is annotated dBu or dBvu. So any time you see dBu, you are dealing with analogue audio levels, and any time you see dBFS, you are dealing with digital audio levels. Also, most FS meters are peak meters, while the dBu scale is used for judging average analogue levels (and it needs to be calibrated to your gear! VU Meter calibration is VITAL).

This is a completely different Beast than a Digital Full Scale Meter.

 The Digital meter to the left has a hard upper limit of 0dB. Anything above that limit will register as awful, loud, crunching, grating noise because there are no bits above the first bit. Make sense? If not, read this post again, then this one

Did you know: almost every piece of analoge gear has a different absolute upper limit!!? For example, in my Focusrite 2i2, the clip level of the analog microphone inputs is +4dbu, but the clip level for the analogue line inputs is +20dBu! And the clip level for the analogue outputs is +10dBu. How confusing is that?!?!

This means that a signal from a microphone that registers more than 4dBu above 0dBu is going to hard clip the microphone preamp. Analogue Hard clipping just means that significant distortion will be added to the sound, sort of like a digital clipping sound but just a hair more pleasing to the ear. As a side note, it is possible to soft clip analogue gear which gives harmonics and saturation. This can be pleasing to the ear, but you want to be careful soft clipping your monitor outputs because it will smear the frequency spectrum!!

In my Focusrite 2i2, the outputs have more headroom than the inputs. Headroom is a term to describe the amount a signal can be turned up before clipping (it works for both digital and analogue). So in Focusrite 2i2, a digital signal of 0dBFS translates into an analogue output signal of +10dbu

Check it out on the Focusrite Website

So, here is where the rubber meets the road. The absolute best level to record your music at (assuming 24-bit), is line level, which is always 0dbu.

On the mic input, we have 4dbu of headroom before clipping. This means that going in through a microphone, +4dbu will translate to 0dBFS. We want to stay WELL below that level. At least 4dBu below so as not to clip the preamp, but at least 10dBu below so as not to clip the outputs.

The best bet? Record at the lowest acceptable volume for the input and output stage. Let's start by using the outputs as our guide for recording something with a microphone, since doing so will keep us well under the clip range of the microphone preamps.

Here's the math: 

if +10dBu = 0dBFS,  then

     0dBu = -10dBFS.

Just subtract the maximum analogue level from both sides of the equation and voila, you have a good peak level to watch on the digital audio scale as you record. This formula works with any recording interface by the way as long as you know what the maximum dBu level is!

What if you are recording something on the 2i2 using the line input? Well, since the line input has way more headroom, the math changes a bit (but not much). With the line level input, +20dBu = 0dBFS. so ....

if +20dBu = 0dBFS, then

       0dBu = -20dBFS

To keep things simple, I record, mix, and master at average level of below -15dBFS. I use an RMS meter to judge this, but you can also just watch the digital peak levels bounce in a DAW and eyeball what the average level might be. This means that as I add more elements to a mix and the levels start to push above -15dBFS, I either select all the faders and pull them all down in unison until I'm back in the sweet spot. Or I just pull the master fader down.

Why is it important to record this way? The reason is simple: because 0dBu is the sweet spot for all analogue gear, including the analogue inputs and outputs of every digital recording interface ever made!! It's also the sweet spot for MANY plugins!!

Its just that figuring out what 0dBu means in terms of the Digital scale can be tricky, but hopefully now you have a better understanding of  how to do that and what's going on behind the scenes. And just like gear, plugins have their own maximum levels, sometimes expressed as dBu. It's insane, but it's worth figuring this stuff out so that you can keep as clean a signal path throughout your recording as possible. The results on each track can be subtle, but across an entire mix it makes a huge difference.

Here's to your recordings!

Ryan

Disclaimer: I do not work for focusrite nor am I advertising for them. I just like the gear of theirs that I've tried. It's great bang for your buck and is very high quality IMHO.

All Images via Wikipedia

Limiters

I use limiters in mixing. I do.

Mostly, I use them to do the job of a compressor when compression isn't cutting it.  I use them to tame uneven vocal performances, I use them to solidify uneven bass lines


But I've recently been challenging myself to listen to the character and sound of the limiters I use, and it's been very enlightening!

The first thing I've noticed is that the expensive L2 limiter from Waves is ... terrible. I used to use it as a staple, but I am now on the hunt for new, transparent limiters.

In the meantime, I love free software ... and a lot of the work I do relies on free VST plugins because I can't yet afford to make and manage subscriptions to big software purchases. I need to eat, after all!

So here is a list of what I believe to be the best of the best Free VST limiters. I'm still pushing the developers of reaper to create a transparent limiter for Reaper. There are a lot of JS limiters, but they all have a character sound of compression.

FREE Limiters
LoudMAX - http://loudmax.blogspot.ca/
Classic Master Limiter - http://www.acoustica.com/plugins/vst-directx.htm
TLS Pocket Limiter - http://hem.bredband.net/tbtaudio/archive/newtbtvstplugins.htm
W1 Limiter (a perfect clone of Waves L1) - http://www.yohng.com/software/w1limit.html