Saturday, September 28, 2013

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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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.

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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,

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