📄 layer-3 faq.htm
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information") and parts that are less sensitive ("data of spectral components").<BR>
All Layers support the insertion of programm-associated information ("ancillary data") into their audio
data bitstream.<BR>
All Layers may use 32, 44.1 or 48 kHz sampling frequency.<BR>
All Layers are allowed to work with similar bitrates:</FONT>
<UL>
<LI><FONT FACE="Arial, Helvetica">Layer-1: from 32 kbps to 448 kbps</FONT>
<LI><FONT FACE="Arial, Helvetica">Layer-2: from 32 kbps to 384 kbps</FONT>
<LI><FONT FACE="Arial, Helvetica">Layer-3: from 32 kbps to 320 kbps</FONT>
</UL>
<P><FONT FACE="Arial, Helvetica">The last two statements refer to MPEG-1; with MPEG-2, there is an extension for
the sampling frequencies and bitrates (see below).</FONT>
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<TD COLSPAN="2"><FONT FACE="Arial, Helvetica"><B>Q: What are the main differences between the three Layers, from a global view?</B></FONT></TD>
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<TD><FONT FACE="Arial, Helvetica"><B>A:</B> From Layer-1 to Layer-3, complexity increases (mainly true for the
encoder), overall codec delay increases, and performance increases (sound quality per bitrate).</FONT>
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<TD COLSPAN="2"><FONT FACE="Arial, Helvetica"><B>Q: What are the main differences between MPEG-1 and MPEG-2 in the audio part?</B></FONT></TD>
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<TD><FONT FACE="Arial, Helvetica"><B>A:</B> MPEG-1 and MPEG-2 use the same family of audio codecs, Layer-1, -2
and -3. The new audio features of MPEG-2 are a "low sample rate extension" to address very low bitrate
applications with limited bandwidth requirements (the new sampling frequencies are 16, 22.05 or 24 kHz, the bitrates
extend down to 8 kbps), and a "multichannel extension" to address surround sound applications with up
to 5 main audio channels (left, center, right, left surround, right surround) and optionally 1 extra "low
frequency enhancement (LFE)" channel for subwoofer signals; in addition, a "multilingual extension"
allows the inclusion of up to 7 more audio channels.</FONT>
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<TD COLSPAN="2"><FONT FACE="Arial, Helvetica"><B>Q: Is this all compatible to each other?</B></FONT></TD>
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<TD><FONT FACE="Arial, Helvetica"><B>A: </B>Well, more or less, yes - with the execption of the low sample rate extension. Obviously,
a pure MPEG-1 decoder is not able to handle the new <I>half</I> sample rates.</FONT></TD>
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<TD COLSPAN="2"><FONT FACE="Arial, Helvetica"><B>Q: You mean: compatible!? With all these extra audio channels? Please explain!</B></FONT></TD>
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<TD><FONT FACE="Arial, Helvetica"><B>A:</B> Compatibility has been a major topic during the MPEG-2 definition phase.
The main idea is to use the same basic bitstream format as defined in MPEG-1, with the main data field carrying
two audio signals (called L0 and R0) as before, and the ancillary data field carrying the multichannel extension
information. Without going further into details, two terms should be explained here: "forwards compatible":
the MPEG-2 decoder has to accept any MPEG-1 audio bitstream (that represents one or two audio channels) "backwards
compatible": the MPEG-1 decoder should be able to decode the audio signals in the main data field (L0 and
R0) of the MPEG-2 bitstream "Matrixing" may be used to get the surround information into L0 and R0: L0
= left signal + a * center signal + b * left surround signal R0 = right signal + a * center signal + b * right
surround signal Therefore, a MPEG-1 decoder can reproduce a comprehensive downmix of the full 5- channel information.
A MPEG-2 decoder uses the multichannel extension information (3 more audio signals) to reconstruct the five surround channels.</FONT>
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<TD COLSPAN="2"><FONT FACE="Arial, Helvetica"><B>Q: In your footnotes, you indicate the use of some "non-ISO" extension
inside your Fraunhofer codec, called "MPEG 2.5", to further improve the performance at very low bitrates
(e.g. 8 kbps mono). What do you mean by this?</B></FONT></TD>
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<TD><FONT FACE="Arial, Helvetica"><B>A:</B> Oh, yes. Well, the MPEG-2 standard allows bitrates as low as 8 kbps,
for the low sample rate extension. At such a low bitrate, the useful audio bandwidth has to be limited anyway,
e.g. to 3 kHz. Therefore, the actual sample rate could be reduced, e.g. to 8 kHz. The lower the sample rate, the
better the frequency resolution, the worse the time resolution, and the better the ratio between control information
and audio payload inside the bitstream format. As the MPEG-2 standard defines 16 kHz as lowest sample rate, we
introduced a further extension, again dividing the low sample rates of MPEG-2 by 2, i.e. we introduced 8, 11.025,
and 12 kHz - and we named this extension to the extension "MPEG 2.5". "Layer-3" performs significantly
better with 8 kbps @ 8 kHz or 16 kbps @ 11 kHz than with 8 or 16 kbps @ 16 kHz.</FONT>
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<TD COLSPAN="2"><A NAME="aboutLayer3"></A><FONT SIZE="3" FACE="Arial, Helvetica"><B>Advanced Features of Layer-3 - or: Why does Layer-3 perform so well?</B></FONT></TD>
<TD> </TD>
</TR>
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<TD COLSPAN="2"><FONT FACE="Arial, Helvetica"><B>Q: Well, I read your statement about "CD-like" performance, achieved
at a data reduction of 4:1 (or 384 kbps total bitrate) with Layer-1, 6..8:1 (or 256..192 kbps total bitrate) with
Layer-2, and 12..14:1 (or 128..112 kbps total bitrate) with Layer-3. Can you explain a little further?</B></FONT></TD>
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</TR>
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<TD><FONT FACE="Arial, Helvetica"><B>A:</B> Well, each audio Layer extends the features of the Layer with the lower
number. The simplest form is Layer-1. It has been designed mainly for the DCC (Digital Compact Cassette), where
it is used at 384 kbps (called "PASC"). Layer-2 has been designed as a trade-off between complexity and
performance. It achieves a good sound quality at bitrates down to 192 kbps. Below, sound quality suffers. Layer-3
has been designed for low bitrates right from the start. It adds a number of "advanced features" to Layer-2:
the frequency resolution is 18 times higher, which allows a Layer-3 encoder to adapt the quantisation noise much
better to the masking threshold only Layer-3 uses entropy coding (like MPEG video) to further reduce redundancy
only Layer-3 uses a bit reservoir (like MPEG video) to suppress artefacts in critical moments and Layer-3 may use
more advanced joint-stereo coding methods</FONT>
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<TD COLSPAN="2"><FONT FACE="Arial, Helvetica"><B>Q: I see. Now, tell me more about sound quality. How do you assess that?</B></FONT></TD>
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<TD><FONT FACE="Arial, Helvetica"><B>A:</B> Today, there is no alternative to expensive listening tests. During
the ISO-MPEG process, a number of international listening tests have been performed, with a lot of trained listeners.
All these tests used the "triple stimulus, hidden reference" method and the "CCIR impairment scale"
to assess the sound quality. The listening sequence is "ABC", with A = original, BC = pair of original
/ coded signal with random sequence, and the listener has to evaluate both B and C with a number between 1.0 and
5.0. The meaning of these values is: 5.0 = transparent (this should be the original signal) 4.0 = perceptible,
but not annoying (first differences noticable) 3.0 = slightly annoying 2.0 = annoying 1.0 = very annoying</FONT>
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<TD COLSPAN="2"><FONT FACE="Arial, Helvetica"><B>Q: Listening tests are certainly an expensive task. Is there really no alternative?</B></FONT></TD>
<TD> </TD>
</TR>
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<TD> </TD>
<TD><FONT FACE="Arial, Helvetica"><B>A:</B> Well, at least not today. Tomorrow may be different. To assess sound
quality with perceptual codecs, all traditional "quality" parameters (like signal-to-noise ratio, total
harmonic distortion, bandwidth) are rather useless, as any codec may introduce noise and distortions as long as
these do not affect the perceived sound quality. So, listening tests are necessary, and, if carefully prepared
and performed, they lead to rather reliable results.<BR>
Nevertheless, Fraunhofer-IIS works on the development and standardisation of objective sound quality assessment
tools, too. And there is already a first product available (contact <A HREF="http://www.opticom.de">OPTICOM</A>), a real-time measurement tool that nicely
supports the analysis of perceptual audio codecs. If you need more information about the Noise- to-Mask-Ratio (NMR)
technology, see our <A HREF="../../../nmr/index.html">NMR-Page</A> or contact <A HREF="mailto:nmr@iis.fhg.de">nmr@iis.fhg.de</A>.</FONT>
</TD>
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<TD COLSPAN="2"><FONT FACE="Arial, Helvetica"><B>Q: O.K., back to these listening tests and the performance evaluation. Come
on, tell me some results.</B></FONT></TD>
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<TD><FONT FACE="Arial, Helvetica"><B>A:</B> Well, for more details you should study one of these <A HREF="http://www.aes.org/index.html">
AES</A> papers or the MPEG documents. For MPEG Layer-3,
the main result is that it always performed superior at low bitrates (64 kbps per audio channel or below). Well,
this is not completely surprising, as MPEG Layer-3 uses the same tool set as Layer-2, but with some additional advanced
coding features that all address the demands of very low bitrate coding. One impressive example is the ISO-MPEG
listening test carried out in September 94 at NTT Japan (doc. ISO/IEC JTC1/SC29/WG11 N0848, 11.Nov. 94). Another
interesting result is the conclusion of the task group TG 10/2 within the ITU- R, which recommends the use of low
bit-rate audio coding schemes for digital sound-broadcasting applications (ITU-R doc. BS.1115).</FONT>
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<TD COLSPAN="2"><FONT FACE="Arial, Helvetica"><B>Q: Very interesting! Tell me more about this recommendation!</B></FONT></TD>
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</TR>
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<TD><FONT FACE="Arial, Helvetica"><B>A:</B> The task group TG 10/2 finished its work in 10/93. The recommendation
defines three fields of broadcast applications and recommends Layer-2 with 180 kbps per channel for distribution
and contribution links (20 kHz bandwidth, no audible impairments with up to 5 cascaded codec), Layer-2 with 128
kbps per channel for emission (20 kHz bandwidth), and MPEG Layer-3 with 60 (120) kbps for mono (stereo) signals for
commentary links (15 kHz bandwidth).</FONT>
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<TD COLSPAN="2"><FONT SIZE="3" FACE="Arial, Helvetica"><B>Q: Where can I find more information?</B></FONT></TD>
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<TD><FONT FACE="Arial, Helvetica"><B>A:</B> For around 10 years, perceptual audio coding is a permanent topic at various scientific
conferences; e.g., the <A HREF="http://www.aes.org">AES</A> (Audio Engineering Society) organizes two <A HREF="../../../events/index.html">conventions</A> per year.
You may find the following papers helpful:</FONT></TD>
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<OL>
<LI><FONT FACE="Arial, Helvetica">Brandenburg, Stoll, et al.: "The ISO/MPEG-Audio Codec: A Generic Standard
for Coding of High Quality Digital Audio", 92nd AES, Vienna Mar. 92, pp. 3336; revised version ("ISO-MPEG-1
Audio: A Generic Standard...") published in the Journal of AES, Vol.42, No. 10, Oct. 94</FONT>
<LI><FONT FACE="Arial, Helvetica">Eberlein, Popp, et al.: "Layer-3, a Flexible Coding Standard", 94th
AES, Berlin Mar. 93, pp. 3493 3) Church, Grill, et al.: "ISDN and ISO/MPEG Layer-3 Audio Coding: Powerful
New tools for Broadcast and Audio Production", 95th AES, New York Oct. 93, pp. 3743</FONT>
<LI><FONT FACE="Arial, Helvetica">Grill, Herre, et al.: "Improved MPEG-2 Audio Multi-Channel Encoding",
96th AES, Amsterdam Feb. 94, pp. 3865</FONT>
<LI><FONT FACE="Arial, Helvetica">Witte, Dietz, et al.: "Single Chip Implementation of an ISO/MPEG Layer-3
Decoder", 96th AES, Amsterdam Feb. 94, pp. 3805</FONT>
<LI><FONT FACE="Arial, Helvetica">Herre, Brandenburg, et al.: "Second Generation ISO/MPEG Audio Layer-3 Coding",
98th AES, Paris Feb. 95</FONT>
<LI><FONT FACE="Arial, Helvetica">Dietz, Popp, et al.: "Audio Compression for Network Transmission",
99th AES, New York Oct. 95, pp. 4129</FONT>
<LI><FONT FACE="Arial, Helvetica">Brandenburg, Bosi: "Overview of MPEG-Audio: Current and Future Standards
for Low Bit-Rate Audio Coding, 99th AES, New York Oct. 95, pp. 4130</FONT>
<LI><FONT FACE="Arial, Helvetica">Buchta, Meltzer, et al.: "The WorldStar Sound Format", 101st AES, Los
Angeles Nov. 96, pp. 4385</FONT>
<LI><FONT FACE="Arial, Helvetica">Bosi, Brandenburg, et al: "ISO/IEC MPEG-2 Advanced Audio Coding", 101st
AES, Los Angeles Nov. 96, pp. 4382</FONT>
</OL>
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<TD><FONT FACE="Arial, Helvetica">Please note that these papers are not available electronically. You have to order
the preprints ("pp. xxxx") directly from the AES.</FONT>
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