Tutorial - I.Basics
Part 1 - Digital Audio - Part 1
What is digital audio made of ?
Digital audio is simliar to pictures taken with a digital camera. It's a bunch of snapshots.
Let's examine the audio material on an audio CD (also called CDDA-Compact Disc Digital Audio).
Here we find two channels of 44100 snapshots per second. These snapshots are called samples and in this case are stored as Pulse Code Modulation data (PCM). Each sample on an audio CD is made of 16 bits, which refers to the bit resolution or bit depth of the PCM data.
16 bits means that each sample can take on any of 2^16 different values. A binary representation of a value might look like this:
1010 1101 1101 1101
So this sample can take on values from 0 to 65535 (binary 1111 1111 1111 1111).
The more values that the audio can take on, the more dynamic range you can hear.
How does audio get digitized ?
This is where a part of your soundcard comes in. It is the Analog-to-Digital Converter (ADC).
It interprets an electrical signal and assigns an absolute value to it, which it writes to memory, so that software can read and store it away in further places, such as your RAM or harddisk. The chain of software your audio data passes through may be
Analog-to-Digital Converter -> RAM -> soundcard driver -> recording software
Playback of digital audio uses a Digital-to-Analog Converter (DAC). This takes the sample and sets a certain voltage on the analog outputs to recreate the signal, that the Analog-to-Digital Converter originally took to create the sample. The DAC does this as faithfully as possible and the first CD players did only that, which didn't sound good at all. Nowadays DACs use Oversampling to smooth out the audio signal. The quality of the filters in the DAC also contribute to the quality of the recreated analog audio signal. The filter is part of a multitude of stages that make up a DAC.
The standard file formats for PCM audio
The most used formats today are :
| WAV |
Can store almost any format via Windows audio codecs,
and therefore can act as a wrapper around many other formats.
Most often used to store 8,16 & 24 bit PCM data of sample rates of 2 kHz to 192 kHz.
16 bit PCM is stored in Little Endian Mode. |
| AIFF |
The WAV equivalent of the Apple Macintosh computers.
Most often used to store 8,16 & 24 bit PCM data of sample rates of 2 kHz to 192 kHz.
16 bit PCM is stored in Big Endian Mode. |
| Sound Designer II |
Used only on Macintosh computers and in large part in a
Protools enviroment, which is a professional audio recording and editing solution.
This format stores its audio data (PCM only) in the data fork and the format information,
such as sample rate, bit depth and mono/stereo choice, in the other fork.
On Macs there are always these two parts to every file. They don't usualy use
filename extensions like *.wav.
Protools can only work with mono SDII(single channel) files, thus it distinguishes
stereo or multichannel files by adding little extensions like .L and .R for stereo files.
5.1 file groups get the suffixes .L(left-front), .C(center), .R(right-front),
.SL(surround left), .SR(surround right) and .LFE(low frequency effects).
Audacity won't actually write this format, but you can read them via the Import Raw Data
function in the Project menu.
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For details on what audio format Audacity can import from and export to, please check
out the Fileformats page of this documentation. Please remember that MP3 is
not a format that stores its audio in PCM form. PCM data is uncompressed.
Next part - Rules of Audacity
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