Labels

BASS (50) COMPRESSION (32) DRUMS (45) EFFECTS (49) EQUALIZATION (30) GUITAR (112) HOME RECORDING (95) IMPULSES (21) INTERVIEWS (19) KARAOKE (1) LIVE (10) MASTERING (61) MIDI (21) MIXING (179) REVIEWS (156) SAMPLES (69) SONGWRITING (19) SYNTH (3) VOCALS (31)

Saturday, September 21, 2013

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.


Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Saturday, September 14, 2013

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:


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!

Saturday, September 7, 2013

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

Saturday, August 31, 2013

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


Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Saturday, August 24, 2013

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 :)

Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Saturday, August 17, 2013

REVIEW: IBANEZ V72Ce

Hello and welcome to this weeks'article!
Today we're going to review an interesting budget acoustic guitar, produced by Ibanez: the V72CE.
These are the cheapest acoustic guitars in the Ibanez catalog, and are single-cutaway dreadnought guitars.
Dreadnought guitars takes the name from a large english battleship, and this name stands for instruments with a larger body than most of its contemporary guitars, and with a deeper, louder tone.




These guitars feature an interesting in-built preamp and tuner, called AEQ200T
The preamplifier has three faders: Bass, Treble and Volume, and obviously an output jack, good to go on a guitar amp or straight to the P.A.
The direct output of this guitar was used to record the sample that can be listened on our article about How to Record and Mix an Acoustic guitar.




Tech Specs (taken from the Ibanez Website):


body shape: Cutaway Dreadnought body
top: Spruce top
back & sides: Mahogany back & sides
neck: Mahogany neck
pickup: Ibanez Under Saddle pickup
preamp: Ibanez AEQ200T preamp w/Onboard tuner
tuning machine: Chrome Die-cast tuners
bridge pins: Ibanez Advantage™ bridge pins
bridge & fretboard: Rosewood bridge and fretboard




Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Thursday, August 8, 2013

HOW TO RECORD A STRING ENSEMBLE!


Hello and welcome to this week's article! Today we're going to talk about a very particular topic, that is not often encountered in a home recording environment but which deserves an in-depth analysis, nonetheless:
How to record a string ensemble? And more in general, how to record an acoustic ensemble that we're going to insert in our mix, with lead and/or ambience role?

I had the opportunity to record a string sextet (violins and cello) called to beef up a rock song, and its role was to give a realistic feel to what before was a synth strings part, so I had to deal with a complete equalization spectrum that needed to be recorded in as few tracks as possible (to close-mike any single element would have been quite useless).


Surfing the web there are many ways to microphone an acoustic ensemble, especially a string one, but here's my own way, developed by experimenting different mic placements.
I have placed two micro-condenser microphones, of the kind often used to track the overhead cymbals of a drumset, with the X/Y method, in order to avoid phasing issues, and each microphone was pointing to the center of its half of the ensemble, trying to place them as symmetric as possible.
Then I had the players with the cellos to sit as centered as possible, and the violins, which are higher pitch instruments, on the sides (remember, the lower frequencies should be as mono as possible, while the higher ones can be placed wherever we want, as explained in this article).

Finally, I have placed a Neumann large-condenser microphone at the center of the ensemble, at the height of the "F-holes" of the Cellos (no pun intended), which are located much lower in height than the violins, in order to make it catch specifically the sound coming from these instruments. 
The idea, as you may have already guessed, is to create a frequency separation between lows and highs for panning purposes, and to add some reverb and "room" to the sound catched from the panoramic microphones, since the ensemble has been recorded on a "dead" room, so there was almost no natural reverb, which is actually very useful when recording acoustic instruments.

Now we should have 3 tracks: left, right and center, which are ready to be mixed!

Become fan of this blog on Facebook! Share it and contact us to collaborate!!