Old 02-18-2008, 12:08 AM   #1
inthepipeline
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Default How well do you know you 1s and 0s

I hope that I'm not going to be in for too much criticism here. It's just that the other day someone asked me a question about digital audio and I couldn't answer it.
It was a really simple question and I should have been able to, but I couldn't. I'd simply forgotten the answer.

I decided that I should do some reading to get myself in touch with things again, so I picked up "Mastering Audio" by Bob Katz. An invaluable work in my opinion.

The in-depth workings of digital audio had been something that I'd paid sparse attention to.
My reason was the usual one: I just want to get on with making music.

The truth is, for me number-crunching isn't any fun, wheras making music is, but the whole number thing is really important in terms of getting the best out of our recordings.

I thought it might be a good idea to start a thread to help anyone who, like me, hates numbers, loves music, loves recording, wants to get the best out of it and would like some practical answers to questions which usually get mathematical ones.

I don't mean to belittle anyone here. There are some great minds in this forum, but I thought it might be an idea to start with some of the more simple stuff and expand from there.
OK, I've written a couple of Q?As here. They are intended to be a compromise between technical accuracy and simplicity, oh and a little provocative perhaps!


Questions about digital audio that you were afraid to ask:

Q.

What are bit Depth and Sample Rate?

A.

OK, Easy. In the computer domain the sound gets drawn like a graph. Imagine you have drawn the shape of a wave across a piece of graph paper and marked all the places where the wave intersects with the grid. The lines from top to bottom represent a value for the magnitude of the sound (bit depth) and the lines across represent the duration or time (sampling rate). These values get recorded and stored. Now get another piece of paper the same and put dots on the paper from the information that was stored. A copy of the wave can be created.
The more squares there are on the graph, the more accurately the original image can be re-drawn, but more information (more points on the graph) need to be recorded, so the file is larger. It contains more information.
When a sound is recorded by a computer, the sound card converts a voltage into a number which is then stored with a time position.
When a computer informs the sound-card to spit out a voltage at a given time it creates a waveform similar to the one which was recorded.

The wave which comes out is a clearer picture of the wave that went in when the picture is taken more accurately (more possible values and more samples over time).

Think of it like digital photography. The more pixels the image has, the sharper it appears.


Q.

CDs are 16Bit, 44.1 Khz. That sounds OK. Should I use this bit depth and Sample rate for recording?

A.

No, not if you can help it!
16Bit, 44.1 Khz was chosen by the music industry as a good compromise between quality and file size. It was considered to be "just good enough to get away with", so to speak.
When we do recordings on our DAWs there are a few reasons to use higher sampling rates and bit depths which are really easy to understand and others which become easy to understand once the easier ones are understood. It's not rocket-science, but they use it!

Going back to the idea of a graph drawn on a piece of paper with the dots marked on it, the original drawing has a nice smooth line which passes through the dots, but the computer only sees the numbers and re-creates the dots, so the soundcard will be inclined to step between these values rather than sweeping through them in the way they were originally put in. They have been given a quantity and then the quantity has been used to attempt to "draw" the wave again. This distortion is known as quantisation distortion and leads to the cold, metallic sound often associated with digital recordings.

If a sound-file is processed within a computer (this could be changing the level, altering the EQ or anything else) the image of the wave will be re-drawn many times in the process and will be changed in a similar way to processing a photo. After altering the image, it will become less and less like the original. The better the image is to begin with, the more we are able to process it before it becomes badly altered.

Choose the best (highest) sample rate and bit depth your sound card can handle. Hard drives are cheap compared to analog tapes. Your material is worth recording well. Don't choose to save a small amount of money to keep your projects compact. Using big file sizes is a good way to store your hard work. You can do more processing of your material before it will deteriorate. You will get a better sound compared to using smaller file sizes.

Q.

How do I set up my recording levels?

A.

Not to high.

Recording with high recording levels is likely to cause clipping (distortion). If files are recorded using a bit depth of 16 you will be far more limited in the amount of flexibility available in your recording levels. When the level hits the top it will clip. With higher bit depths there are a greater number of values available for the wave of your recording being drawn on the graph, so when the levels are set, they can be much lower without creating as much distortion.

The graph paper again:
Let's suppose that when the waveform was originally drawn, it was drawn on directly onto the graph, but onto a transparency and the transparency laid over the graph paper. Suppose that we changed the paper for a much larger piece. The wave is the same size, but the amount of room between the largest value of the wave has changed. That space is known as headroom. The greater the bit-depth, the more headroom you have available. This doesn't mean that you can overload the pre-amps on your soundcard, but it does mean that there is no need to try and record everything as loud as possible.
There is no point in recording "too hot".

Q.

My sound card can only record 24bit. What is the advantage of recording 32bit float?


A.

Simple. Your recording was recorded in that great big file. You had all those extra values to play with when you recorded it. Now it's time to play. You can equalise, compress, expand, gate, mutilate and as much as the file will eventually deteriorate, it won't do it with as few operations and it will be far more difficult to make it clip (almost impossible). This is where working with digital sound really gets to be fun. It's still important not to be abusive with the levels at this point, but you will have much more flexibility than if you were to use good ol' fashion 16 bit, 20 or even 24.

When the computer records a 32bit/float file from your soundcard, the values your card created to draw the waveform remain as they were, occupying a small space on that huge graph, which can now be manipulated with great flexibility. When the computer outputs back to your card, you will hear 24bit. the additional accuracy is recorded, but not output. However, in the meanwhile it has been possible to do plenty of manipulation of the waveform without as much damage to the image. So, if you like, you have a deliberately distorted image which is sharper.

Of course, CPU overhead has also to be considered. The more information to be processesed, the higher the CPU load, but then with careful planning and rendering your tracks at the right time, your imagination can become your boundary. If it isn't I'd like to work with you....but you would probably scare me!
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