| This section explains briefly the
basic topics related to digital audio technology.
What is CD-DA or DAE? (Back
to top)
The process of direct reading the audio data is called CD
Digital Audio (CD-DA) or sometimes referred to as "Digital
Audio Extraction" (DAE) or as "CD ripping"
. Lavavo CD Ripper uses CD-DA to read the audio data of the
audio CD disc and write the resulting data into a WAV file
or a MP3 file. The advantage of CD-DA above recording the
file using your sound card is that there is no loss of music
quality; the data in the WAV file is an exact copy of the
data on the disc. (In technical terms, using your sound card,
the digital data from the audio CD is converted to an analog
signal, which is then digitized again by the sound card. Often
the sound quality will deteriorate when going this route,
and producing noisy WAV files.)
Furthermore, DAE is also a lot faster than ripping audio
data with your sound card.
The purpose of audio compression
(Back to top)
Until audio compression was invented, high-quality digital
audio data took a lot of hard disk space to store. Let's go
through a short example. Suppose you want to sample your favorite
1-minute song and store it on your hard disk. Because you
want CD quality, you sample at 44.1 kHz, stereo, with 16 bits
per sample.
44100 Hz means that you have 44,100 values per second coming
in from your CD-ROM (or input file). Multiply that by two,
because you have two channels, left and right. Multiply by
two again because you have two bytes per sample (i.e. 16 bits
per sample). The song will take up 44100 samples/s x 2 channels
x 2 bytes/sample x 60 seconds/min Mbytes equals about 10 megabytes
of storage space on your hard disk, for each minute!
If you want to download that over the Internet, given an
average 28.8 modem, it would take you about 10,000,000 bytes
x 8 bits/byte / (28800 bits/s) x (60 s/min) or about 45 minutes.
Just to download one minute of music!
Digital audio coding (also called "digital audio compression")
is the art of minimizing storage space (or channel bandwidth)
requirements for audio data. Modern perceptual audio coding
techniques (like MPEG, Vorbis OGG, AAC) consider how we actually
perceive sound, to give higher fidelity to the parts of the
sound that we hear most clearly, and to simplify the parts
of the sound where we are less likely to notice any difference.
Using compression that uses these techniques can take uncompressed
audio, and represent it in about a twelfth the original disk
space, without any noticeable loss in fidelity. It can compress
it even further and further, if you are willing to put up
with less and less fidelity.
So such schemes are the main technology for high quality
low bit-rate applications, like soundtracks for CD-ROM games,
solid-state sound memories, Internet audio, digital audio
broadcasting systems, and the like.
When you deal with compression technology, often you have
to specify the "bitrate". The "bitrate"
refers to how much data is needed to represent how much sound.
If less data represents each second of sound, then the sound
quality will be lower; but if more data represents each second,
the quality will be better.
For example, an MP3 encoded "at a bitrate of 128
Kbps" (a very common compression rate) represents
each second of sound using 128,000 bits -- roughly like
the quality of FM radio with good reception. The same
sound encoded "at a bitrate of 64 Kbps" will represent
each second using half as much data, so the file on the whole
will be half the size of the 128 Kbps and the audio quality
will be lower. It's not quite right to say that it's "half
as good", since it's hard to really use numbers for how
good we think something sounds; but 64 Kbps could be compared
to the quality of a copied cassette tape.
Which bitrate should I use to get
CD-Quality? (Back to top)
This is another of those questions that people debate endlessly.
There really is no single (correct) answer - it depends on
your ears, your equipment, the quality and complexity of the
input sound data, etc. I think it's probably fair to say that
128 Kbps is a bit on the low side for music, but it may sound
fine from some encoders for some songs under some circumstances;
while 160 Kbps done by a good encoder should sound pretty
good to most people under most circumstances. But I know some
people will insist on higher standards. Which MP3 encoder
you use really does make a difference - I personally recommend
using the LAME MP3 encoder (the default in Lavavo CD Ripper),
or the Fraunhofer MP3 encoder.
Also note that simple speech is generally much less acoustically
complex than music, and that people generally care less about
the audio quality of speech, as long as it's understandable.
This means that bitrates below 128 Kbps can be just fine for
compressing speech, even though those bitrates are usually
not even worth considering for music.
Bitrate Options (Back
to top)
There are three types of bitrate options that you can specify
for each the encoder (although some encoders may not allow
any options).
1) Constant Bitrate (CBR)
This is the default encoding mode, and also the most basic.
In this mode, the bitrate will be the same throughout the
whole file. So, a second of audio from one part of the file
takes just as much disk space as a second from any other part
of that file - regardless of whether either part is silence,
acoustically simple, or quite complex. This means that you
are likely to hear distortion more in the complex parts than
in the simple parts. The advantage of CBR formats is that
even older players understand them, and that you can reliably
predict the file size from the duration of the sound (or vice
versa).
2) Average Bitrate (ABR)
In this mode, you tell the encoder to aim for an average
bitrate that you specify, skimping on the simpler parts of
the music, and using higher bitrates for the parts of your
music that are more complex. The result will be of higher
quality than you'd get in a CBR encoded file of the same size.
This mode is highly recommended over CBR. This encoding mode
is similar to VBR.
3) Variable bitrate (VBR)
In this mode, you say what level of quality you want in the
output file, and the encoder compresses each second as best
it can to get just that level of quality - using less information
to represent simpler parts of the song, and more information
to represent the more complex parts. However, this mode relies
heavily on the encoder's model of how you perceive quality,
and could lead to a few "bad choices" in the encoding
process. If possible, you may want to specify a minimum bitrate
(e.g. 64 Kbps) to avoid those potential errors.
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