REFERENCING and MIXING!

Hello and welcome to this week's article!
This article can be considered an extra part of our Ear training section, and explains practically how to learn something about a professional record's song and to translate it in our mix.
Let's picture this scenario: we have our mix project in our Daw, we add a new stereo track and import our favourite song, lowering its volume until it matches the one of our project.
Now we can investigate that song, asking it the typical questions that we have already seen on the second and last part of our ear training aricle:

"how much room is reserved for rhythm guitars? Where is located the snare? How deep the bass goes? how much room is taken from the vocals?" ...and so on, and we use the tools and the methods explained in those articles to find out how our favourite mix is composed.

Once we have clear in mind how the song is constructed, tonally speaking, we need to compare it with ours, keeping in mind that 60% of the tone shaping is in the recording, and that we cannot twist a sound into a completely different one without runining it: we can cut frequences doing relatively little damage to the nature of the sound, but when boosting, the more we boost, the more it will sound unnatural and digitally warped (as it actually happens).

Comparing our song with the referincing one we coud notice for example that our snare drum sound it's completely different from the imported one: that one for example can be high oriented, because the drummer had a small snare with a very high tuning, while ours is dark because the snare was huge and low tuned, so the sound will never be similar, and it will sit on a totally different place in the mix. We cannot completely transform it because it's gonna sound awful, so we got two options: replace our snare with a sample through a drum replacer, or compare our track with another one that has sounds more similar to ours. It has no sense to have for example a grunge song to mix, and compare it with a death metal one only because we like the snare: it will only confuse us and lead us to wrong choices.

What it's particularly interesting is to understand how the audio engineers have treated the 2 most problematic areas: the one in which the vocals and the snare drum lies, and the low end.
The first one it's the zone the human ear is most sensible to, so we can understand by listening to it how a mix is balanced: are the vocals at the same volume of the drums, as in the modern rock?
Are kick and snare louder than vocals (as in death metal)?
Or the opposite, as in pop music?
We can apply the same pholosophy to our mix, and it will make the difference.
This area is also interesting to study, to understand how the engineer has treated guitars: has he cut the guitar frequences here to leave room to vocals and snare, or did he let the guitar be prominent, sacrificing frequences of the other instruments on that area?
About the low end: where does the guitar stops and where does the bass pops out?
The lower the guitar tuning is, the more usually a guitar is low-frequency oriented and "eats" room to the bass.
Is the bass distorted? How much?
To the point of becoming just a complement of the guitar as it happens in many black or death metal albums?
Or at the opposite, it's just lows and it lies on a range reserved to it?
How loud the cymbals are? Usually the louder they are, the more the album acquires a "garage", realistic, alternative rock feeling, and the more it becomes "dirty".
In pop music and modern metal, cymbals are held very low in volume, while the indie bands likes them to be high and resonating, so, again, it's a style choice that we must understand and use to our advantage.

These are all lessons in sound balancing and eq choices that we can absolutely learn on our own from the masters, and apply them on our mixes, keeping in mind that in the end our mix has to sound well, to be euphonic and balanced with what we got, that is totally different from the original sounds tracked for the referencing songs.
We cannot be too stubborn in wanting a sound that we cannot obtain because the raw track is too different, and we should focus instead into making great music with what we got.
In the end, that is really all that matters.

You can learn more about the best uses of a reference track HERE.


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BIT DEPTH and SAMPLE RATE! a guide for dummies

Hello and welcome to this week's article!
Today we continue with the technical stuff made easy, and we're going to find out what are and how to use the Bit Depth and the Sample rate.

First off: why do we need them?
Because we need to choose the right setting before starting to record our tracks: the different settings will affect the dimension of each recorded track, its characteristics and the cpu usage of the project itself, so we must find the right compromise between quality and usability, especially when it comes to large projects.
The types of bit depth and sample rate availabe is variable according to the type of Audio Interface, and can reach 24bit and 192khz (as a maximum), but if an audio cd, an mp3 or a Youtube video can read only files at 16bit and 44.1khz, why do we need to record at settings so different?
The idea behind the availability of those settings is that we can record at the maximum fidelity possible, and then Dither the signal down to the standard format, obtaining a result that can be better than recording straight in the cd format.

- Bit Depth: why does people record usually al 24 bit? Because Bits are all about headroom: with 24 bits we're allowed to record hotter signals without the risk of clipping, or to record normal signals with a much quieter "noise floor".
In facts, in the analog era, it was important to record with as much signal as possible before clipping, because that way there was much signal emerging from the noise floor, which was quite high; in the digital domain, especially with 24 bit interfaces, the noise floor is so low that we can perfectly record with conservative levels without risking anything.
Sometimes the DAW offers the opportunity to record at 32 bit, but since 99% of audio interfaces in the market can record at a maximum of 24 bit, this feature should not be used.

- Sample rate: to understand this parameter, think about the frames of a movie. If a movie is 24 frames per second it's good, if it's 60fps it's much more fluid.
Sample rate does a similar thing for the audio section: it controls how many snapshots of an audio wave are taken in a second.
The reality is that 44.1khz are enough to represent well basically any musical style, so unless we have some particular need that forces us to use different settings, 44.1khz it's perfect, and at the same time it doesn't weight too much both on our Cpu and hard drive.
Obviously if you want to record your album at higher sample rates feel free to experiment, and if you want, let us know the differences you've noticed!

Our suggestion for a project is to record at 24bit and 44.1khz, that way there will be enough headroom and a great noise to signal ratio, and at the same time the project will not be too heavy on the cpu and hard drive.
Click here for a dedicated article about why recording at a too high sample rate can be even detrimental to our project.


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ASIO, BUFFER SIZE and LATENCY! A guide for dummies.

Hello and welcome to this week's article!
Today we're going to talk about a topic that comes before the beginning of a recording session: how to set up the drivers in our computer in order to be able to work without problems.

First off the Drivers: each Audio Interface has its own driver software, that "explains" to our computer how to communicate with it.
When it comes to using a Daw, the part of the interface driver that is important is the ASIO section (audio streaming input output): this is the protocol for audio transmission, created by Steinberg, that lets us record and play one or more channels at the same time, with a very low latency.
Once we have set the Asio driver of our interface in our Daw, the workstation will stream all the input and output audio data from our interface.
If we have no Audio Interface but we need to record something on a Daw, we can use a driver called Asio 4 All, keeping in mind that the integrated preamp of our motherboard is really cheap, and it's a good idea to buy a dedicated interface as soon as possible, if we want to produce music.

Now that we have all setup we can try to create or to open a project, and see if we can hear everything well or there are problems: we could experience some mistimed sound, or some random pop or crackle (notice that this errors are not present on the tracks, they are just playback artifacts generated randomly when the Cpu is under an excessive stress).
This happens because our computer, in order to let us hear properly all the tracks (with all the eventual real time processing) needs to buffer the audio prior to let us hear it: the smaller the buffer size is, the lower the latency will be;
making the buffer smaller comes at a price, however.
With a smaller buffer, there is less overhead for delays in processing, therefore the CPU will need to work harder to ensure that any delay is kept within the time allowed by the buffer, and the more the Cpu works and the smaller the buffer is, the more is likely to experience dropouts, pops and crackles.

What we need is a very good Cpu and a lot of Ram, in order to be able to keep the latency low without too many problems; we can start by choosing from the Interface driver a small buffer and as soon as we experience problems we can increase it of one step, until we find the buffer size that lets us work perfectly at the minimum latency possible.

Coming to the latency, we're obviously talking about the delay between when we play a note and when it is received from the computer, processed and sent back to us through the monitors (or the headphones).
When mixing, latency is not a major problem if all the tracks are playing in perfect synchronization among them, the problem is particularly tedious when recording: if we are recording a singer or a guitarist, the maximum monitoring latency tolerable is 10ms or less: with a higher latency it becomes almost impossible to play.
If the latency is unbearable the only solution is to reduce the buffer size, and if this brings pops and crackles, the only thing to do is to try to lighten the Cpu load by disabling all the unnecessary real time effects, or muting tracks, leaving on just the most essential ones.
The ideal is to use a smaller buffer when recording, and a larger one when mixing.


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Post n.100: updating week!

Hello everyone! This week, in occasion of our N.100 POST, instead of publishing a new article I've done some debug to the blog: fixed some links, corrected some mistakes and totally reviewed and updated all the 3 parts of the "Tree" article of this Blog:

Home recording and mixing at zero cost!

This article, which was one of the first of this blog, has been corrected and extended, and now covers every aspect of a recording production. From there you can follow the links and get in depth into the "tree arms" of recording, mixing and mastering. 

Have fun!

REVIEW: Orange AD30!

Hello and welcome to this week's article!

Today we're going to talk about an interestig amplifier, the Orange AD30.

This is a 30w tube amplifier (4 x 12ax7 in the preamp, 4 x el34 in the poweramp section,

plus a gz34 rectifier valve), that comes with two channels: a clean one that can be driven up to becoming an overdrive, and a drive channel that goes from a crunch to a classic british distortion, and at the highest drive levels it can be used for classic metal too.

Obviously the brand is one of those few (along with Marshall) that managed to shape the "british sound" as we know it, and the offer of this brand ranges to almost any kind of music and wattage.
Famous Orange endorsers are: Rush, My Chemical Romance, Cheap Trick, Jimmy Page, Slipknot, Down, Deftones, Kyuss and many others.
We have used this amp for the recording of the clean/od tones of an album, dual microphoning it, and the fatness of the tone was impressive: the power amp was really giving the tone a layer of warmth and roar.

Features (taken from the website):
All valve, foot switchable two channel, valve rectifier,

Controls
Volume, Low, Mid, High, Overdrive, overdrive on/off, Gain

Output Power (Heads and Combos)
30 Watts

Valves (Heads and Combos)
Preamp: 4 x Ecc83/12ax7
Power amp: 4 x EL84
Rectifier: GZ34

Speaker Output Options (Heads)
1 x 16 Ohm cabinet connected to the 16 Ohm output
1 x 8 Ohm cabinet connected to one of the 8 Ohm outputs
2 x 16 Ohm cabinets each connected to one of the 8 Ohm outputs

Unboxed Weight
16.5 KG
36.4 lb

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MID/SIDE PROCESSING (M/S PROCESSING): a guide with free Vst Plugins

Hello and welcome to this week's article!
Today we're going to talk about a certain type of processing that is usually seen in the Mastering Phase, but that can also be handy in various Instrument group Buss: the Mid-Side processing.
This is a type of processing, as you may have guessed, that involves a Stereo Field: there must be on the same track or buss a soundstage with elements panned to the left, some other in the middle, and others to the right.

Mid/Side is a technique in which we have certain plugins that can split the content that comes from the middle of the wave's soundstage from the one coming from the sides, and these plugins can process just the part of the sound we want or both differently; for example we can Compress just the sounds that comes from the sides, or we can eq differently the part of the sound coming from the centre from the one coming from the sides.

The Mid/Side plugins can therefore do any kind of task: the basic idea is that they translate (as you can see from the image) the signal from left-right to mid-side, and process independently these two parts; it is a kind of processing that is not only sometimes needed in mastering, but that is often crucial when we need to tweak a song that has been provided to us just as a stereo track.

Unlike many other effects, it's not easy to find a mid side processor for free.
Here are two of the few ones available:

- Terry West Mid Side Signal Processor: an interesting channel strip with Gain, Equalization and Saturation both for middle and sides.

- Voxengo Marvel Geq: an equalizer with Mid Side functions


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HOW MIC POSITIONING AFFECTS THE SOUND TRANSIENT

Hello and welcome to this week's article!
The topic we're going to talk about today is linked to the Transient article, to the microphoning articles and to the Reverb ones.

Let's make this example: we're on a room, and we have an acoustic instrument, for example the Wind Chimes on the picture.
The first thing we need to understand when we will record them is "will this instrument need to be upfront or in the background of our soundstage?".
According to the answer, we could position our microphone in a different way, especially if we have a good condenser one, very sensitive and capable of catching the details of the room reverb.
What is a Transient? It's the body of a sound.
As we have already seen in the article about transient shapers, transient it's the loudest and most characteristic section of a sound, therefore it is the first one to get ruined when increasing Compression or limiting.

To place a microphone closer or farther from the source will drastically change the transient of a sound: the closer we will put the mike, the more attack we will get, the sound will result more upfront and brighter, but with less body, while setting it in the other side of the room will make the loudest part of the tracked sound the boominess of the room; it will sound as we would hear it from that place: far, in the background.

Transient shaping plugins help us in recovering the attack and the "snap" from a sound recorded with a microphone set too far, or in putting in the background some instrument that is popping out too much in the mix, but this article is to remind you that if the microphonation of the instruments is done correctly, there won't probably be the necessity to use such corrections: the first and best sounding transient shaping tool is the microphone, and our understanding of what is going on in the room:

- how big the room is?

- Has the room had an acoustic treatment to reduce reverbs?

- Is the recorded sound sufficiently bright while mantaining body?

- Are there annoying resonances that needs to be eliminated?

Obviously these questions doesn't have an answer good for every situation and every instrument: some instruments (such as Strings) sounds better in an untreated room since they benefit from natural reverb, some others (drums) can be completely ruined if tracked in a big, untreated room.
Just don't rely only on the plugins to solve this problems, spend some more time in trying to fix them at the source and the whole record will benefit, guaranteed :)

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