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<FONT SIZE="4" FACE="Arial, Helvetica"><B>Basics about MPEG Perceptual Audio Coding</B></FONT></TD>
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<LI><FONT FACE="Arial, Helvetica"><B><A HREF="#1">Introduction</A></B></FONT>
<LI><FONT FACE="Arial, Helvetica"><B><A HREF="#2">The purpose of audio compression</A></B></FONT>
<LI><FONT FACE="Arial, Helvetica"><B><A HREF="#3">The two parts of audio compression</A></B></FONT>
<LI><FONT FACE="Arial, Helvetica"><B><A HREF="#4">How does it work?</A></B></FONT>
<LI><FONT FACE="Arial, Helvetica"><B><A HREF="#5">Compression ratios, bitrate and quality</A></B></FONT>
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<TD COLSPAN="2"><A NAME="1"></A><FONT FACE="Arial, Helvetica"><B>Introduction</B></FONT></TD>
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<TD><FONT FACE="Arial, Helvetica">There is a lot of confusion surrounding the terms <B>audio compression</B>, <B>audio encoding</B>, and <B>audio decoding</B>.
This section will give you an overview what <B>audio coding</B> (another one of these terms...) is all about.</FONT>
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<TD COLSPAN="2"><A NAME="2"></A><FONT FACE="Arial, Helvetica"><B>The purpose of audio compression</B></FONT></TD>
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<TD><P><FONT FACE="Arial, Helvetica">Up to the advent of audio compression, high-quality digital audio data took a lot of hard disk space to store (or channel bandwith to transmit).<BR>
Let us go through a short example. You want to sample your favorite 1-minute song and store it on your harddisk. Because you want CD quality, you
sample at 44.1 kHz, stereo, with 16 bits per sample.<BR>
44.100 Hz means that you have 44.100 values per second coming in from your sound card (or input file). Multiply that by two
because you have two channels. Multiply by another factor of two because you have two bytes per value (that's what 16 bit
means). The song will take up</FONT>
<P><FONT FACE="Arial, Helvetica"><CENTER>44.100 samples/s * 2 channels * 2 bytes/sample * 60 s/min = around 10 MBytes</CENTER></FONT>
<P><FONT FACE="Arial, Helvetica">of storage space on your harddisk. If you wanted to download that over the internet, given an average 28.8 modem, it would take you</FONT>
<P><CENTER><FONT FACE="Arial, Helvetica">10.000.000 bytes * 8 bits/byte / (28.800 bits/s * 60 s/min) = around 49 minutes.<BR>
<BR><B>Just to download one minute of stereo music!</B></FONT></CENTER>
<P><FONT FACE="Arial, Helvetica">Digital audio coding, which - in this context - is synonymously called digital audio compression as well, is the art of minimizing storage
space (or channel bandwidth) requirements for audio data. Modern perceptual audio coding techniques (like MPEG Layer-3 or MPEG-2 AAC) exploit the properties
of the human ear (the perception of sound) to achieve a size reduction by a factor of 12 with little or no perceptible loss of quality.<BR>
Therefore, such schemes are the key 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.</FONT>
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<TD COLSPAN="2"><A NAME="3"></A><FONT FACE="Arial, Helvetica"><B>The two parts of audio compression</B></FONT></TD>
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<TD><P><FONT FACE="Arial, Helvetica">Audio compression really consists of two parts. The first part, called <I>encoding</I>, transforms the digital audio data that resides,
say, in a WAVE file, into a highly compressed form called <I>bitstream</I> (or coded audio data). To play the bitstream on your soundcard, you need the
second part, called <I>decoding</I>. Decoding takes the bitstream and reconstruct it to a WAVE file.</FONT>
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<TD COLSPAN="2"><A NAME="4"></A><FONT FACE="Arial, Helvetica"><B>How does it work?</B></FONT></TD>
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<TD><P><FONT FACE="Arial, Helvetica">Highest coding efficiency is achieved with algorithms exploiting signal redundancies and irrelevancies
in the frequency domain based on a model of the human auditory system.<BR>
All coders use the same basic structure. The coding scheme can be described as "perceptual
noise shaping" or "perceptual subband / transform coding". The encoder analyzes the spectral
components of the audio signal by calculating a filterbank (transform) and applies a psychoacoustic
model to estimate the just noticeable noise-level. In its quantization and coding stage, the encoder
tries to allocate the available number of data bits in a way to meet both the bitrate and masking
requirements.<BR>
The decoder is much less complex. Its only task is to synthesize an audio signal out of the coded
spectral components.<BR>
All Layers use the same analysis filterbank (polyphase with 32 subbands). Layer-3 adds a MDCT
transform to increase the frequency resolution.</FONT>
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<TD COLSPAN="2"><A NAME="5"></A><FONT FACE="Arial, Helvetica"><B>Compression ratios, bitrate and quality</B></FONT></TD>
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<P><FONT FACE="Arial, Helvetica">It has not been explicitly mentioned up to now: What you end up with after encoding and decoding is not the same sound file
anymore: All superfluous information has been squeezed out, so to say. (More precisely: the redundant and irrelevant parts of the sound signal.) The reconstructed WAVE file differs from
the original WAVE file, but it will sound the same - more or less, depending on how much compression had been performed on it.</FONT>
<P><FONT FACE="Arial, Helvetica">Because compression ratio is a somewhat unwieldy measure, experts use the term bitrate when speaking of the strength of
compression. Bitrate denotes the average number of bits that one second of audio data will consume. The usually units here are kbps, which is kbit per second, or 1000 bits/s.</FONT>
<P><FONT FACE="Arial, Helvetica">For a digital audio signal from a CD, the bit-rate is 1411.2 kbps.
With <A HREF="aac/index.html">MPEG-2 AAC</A>, CD-like sound quality is achieved at 96 kbps.</FONT>
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