Article View: cn.bbs.comp.multimedia
Article #5264about MP3
From: Xiao Liu
Date: Tue, 09 Dec 1997 00:00
Date: Tue, 09 Dec 1997 00:00
627 lines
42005 bytes
42005 bytes
Introduction - or: What is "MPEG Audio Layer-3"?
Today, efficient coding techniques are a must for cost-effective processing of digital audio and
video data by computers. Data reduction of moving pictures and sound is a key technology for any
application with limited transmission or storage capacity. In the recent years, a lot of progress has
been achieved. While there (still) exist several proprietary formats for audio and video coding, the
ISO/IEC standardisation body has released an international standard ("MPEG") for powerful audio
and video coding tools (see: Overview about the ISO-MPEG Standard - or: What is MPEG all
about?).
Without data reduction, digital audio signals typically consist of 16 bit samples recorded at a
sampling rate more than twice the actual audio bandwidth (e.g. 44.1 kHz for Compact Disks). So
you end up with more than 1400 kbit to represent just one second of stereo music in CD
quality. By using MPEG audio coding, you may shrink down the original sound data from a CD by
a factor of 12, without losing sound quality. Factors of 24 and even more still maintain a sound
quality that is significantly better than what you get by just reducing the sampling rate and the
resolution of your samples. Basically, this is realized by "perceptual coding" techniques addressing
the perception of sound waves by the human ear (see: Basics of Perceptual Audio Coding - or:
What is the trick?).
Using MPEG audio, one may achieve a typical data reduction of
1:4
by Layer 1 (corresponds with 384 kbps for a stereo signal),
1:6...1:8
by Layer 2 (corresponds with 256..192 kbps for a stereo signal),
1:10...1:12
by Layer 3 (corresponds with 128..112 kbps for a stereo signal),
still maintaining the original CD sound quality.
By exploiting stereo effects and by limiting the audio bandwidth, the coding schemes may achieve
an acceptable sound quality at even lower bitrates. Layer-3 is the most powerful member of the
MPEG audio coding family. For a given sound quality level, it requires the lowest bitrate - or for a
given bitrate, it achieves the highest sound quality (see: Advanced Features of Layer-3 - or: Why
does Layer-3 perform so well?).
Some typical performance data of Layer-3 are:
sound quality
bandwidth
mode
bitrate
reduction ratio
"telephone sound"
2.5 kHz
mono
8 kbps (*)
96:1
"better than shortwave"
4.5 kHz
mono
16 kbps
48:1
"better than AM radio"
7.5 kHz
mono
32 kbps
24:1
"similar to FM radio"
11 kHz
stero
56...64 kbps
26...24:1
"near-CD"
15 kHz
stereo
96 kbps
16:1
"CD"
>15 kHz
stereo
112..128kbps
14..12:1
*: Fraunhofer uses a non-ISO extension of Layer-3 for enhanced performance ("MPEG 2.5")
All in all, Layer-3 is the key for numerous low-bitrate, high-quality sound applications (see:
Applications - or: Layer-3, what is it good for?).
Applications - or: Layer-3, what is it good for?
A key technology like Layer-3 is useful for a pretty large spectrum of applications - practically
almost any system with a limited channel capacity may benefit from it. The following chapters
identify some main areas and list some companies that are actively exploiting the Layer-3
technology. For product-related information, please contact these companies directly.
Music Links via ISDN
Digital telephone networks (ISDN = Integrated Services Digital Network) offer reliable dial-up
links with two 64 kbps data channels per basic rate adapter; other regional networks (in
North-America) use 56 kbps data links. Transmission fees are often rather similar or identical to the
traditional analog phone lines - those allow to transmit up to 28.8 kbps (V.34 modem) or even 32
kbps ("V.34+").
Using Layer-3, a low-cost narrowband ISDN connection allows to transmit CD-quality sound.
Audio professionals, like broadcasting stations and sound studios, benefit from the
"music-by-phone" application in various ways. They save money, as they only pay transmission fees
for the actual time of usage (not 24 h a day in case of a leased phone line) and for a rather small
data channel (one ISDN phone connector for a stereo music link). Radio stations increase the
attractiveness of their programs, as reporters transmit high-quality takes (e.g. an interview) or live
news without annoying "telephone sound". And new applications become possible, e.g. a "virtual
studio", where remote artists may play along some preproduced material, without actually travelling
to the studio.
Examples:
In 1992, Radio FFN, a private broadcasting station in Niedersachsen, Germany, replaced its
leased phone lines with ISDN and Layer-3 codecs, to transmit 8 local programs 20 min per
day to the central broadcasting studio. This move saved them transmission fees of more than
300.000 US$ per year.
As one of the first real-world trials, all private radio stations of Germany very successfully
used Layer-3 codecs during the Winter Olympic Games in Albertville (France) as reporter
links between the various sporting events and their central studio in Meribel.
At the International Music Festival 92 in Bergen, Arne Nordheim composed a piece of
music, where an organ in the church of Trondheim played along with the symphony orchestra
in Bergen; the sound of the organ was transmitted via ISDN and a Layer-3 codec.
Since 1992, various manufacturers are providing equipment ("codecs") for professional audio
applications: AVT, Broadcast Electronics, CCS, Dialog 4, Telos.
Digital Satellite Broadcasting
Pioneered by WorldSpace, a worldwide satellite digital audio broadcasting system is under
construction. Its name is "WorldStar", and it will use three geostationary orbit satellites called
"AfriStar 1" (21 East), "CaribStar 1" (95 West), and "AsiaStar 1" (105 East), with AfriStar 1 being
launched in mid-1998. The other satellites will follow until mid-1999. Each satellite is equipped with
three downlink spot beams that are pointed so as to cover populations that provide the greatest
radio listener base. Each downlink uses TDM (time division multiplexing) to carry 96 prime rate
channels (16 kbps each). The prime rate channels are combined to carry broadcast channels
ranging from 16 kbps to 128 kbps; the broadcast channels are coded using MPEG Layer-3. The
prime rate channels may even be dynamically allocated to meet the demands of the broadcast
service (e.g. 4 channels combined for 1 hour to allow FM quality stereo (64 kbps) for the
transmission of a concert with classic music, followed by 1 hour with 4 separate news channels (16
kbps) in 4 different native tongues).
WorldSpace is offering channels on its three satellites for lease to international and national
broadcasters. Channel reservation agreements already have been signed with a number of major
broadcasters, including Voice of America, Radio Nederland, the Kenya Broadcasting Corporation,
the national broadcasting authority of Ghana, the national broadcasting authority of Zimbabwe,
New Sky Media of Korea, and RCN of Columbia. Nearly 1 billion $ in private financing has been
raised to cover acquisition of the satellites and for most of the operational costs through full system
implementation in 1999. France�s Alcatel Espace is the spacecraft prime contractor and supplies
the telecommunications payload.
The radio receivers will be designed for maximum convenience of use at a minimum cost. Low cost
receiver will use a small compact patch antenna, will require practically no pointing, and will tune
automatically to selected channels. Higher end receivers are also envisioned. In a press release from
5. June 96 (Montreux, Switzerland), WorldSpace declared that it has awarded production
contracts for two million receiver chips; the contracts were issued to SGS-Thomson and ITT
Intermetall, authorizing each company for an initial production of one million receiver chip-sets.
ITT Intermetall has already gained Layer-3 knowhow by using its mask-programmed DSP
technology to develop a single-chip Layer-3 decoder named "MAS 3503 C". This chip supports
only MPEG-1 Layer-3.
Audio-on-Demand
The Internet is a world-wide packet-switched network of computers linked together by various
types of data communications systems. Professional Internet providers usually access the network
through rather high bit-rate links (e.g., primary rate ISDN with 2 Mbps or ATM with up to 2
Gbps). However, the average consumer uses low cost, low bit-rate connections (e.g., basic rate
ISDN with 64 kbps or phone line modems with 28.8 or 14.4 kbps). The actual transmission rate
depends on the current user load and the infrastructure of the part of the Internet in use. From a
client�s point of view, it may unpredictably vary between zero and the maximum bit-rate of its
network modem, with an average bit-rate somewhere in between.
Without audio coding, downloading uncompressed high-quality audio files from a remote Internet
server would result in unfavourably long transmission times. For example, with an average
transmission rate of 28.8 kbaud (optimistic guess), a single 3-min stereo track from a CD (31.7
Mbyte) would require a download time of more than 2 hours. Therefore, audio on the Internet calls
for an audio coding scheme that maintains sound quality as far as possible and allows real-time
decoding on a large number of computer platforms without special add-on hardware. Layer-3 fits
very well into this scenario - real-time players (like WinPlay3) are available. Intranets present an
interesting special case, as they usually provide sufficient bitrate to allow a number of real-time
audio links. Furthermore, our experiments indicate that using the http protocol, a real-time
connection with 56 (112) kbps is possible with one (two) ISDN phone line(s).
If content providers are willing to add audio data onto their Internet servers, they have to consider
carefully the copyright aspects of the music industry (e.g., artists, producers, record companies).
They must not violate these rights by their actions! In the framework of a European project called
MODE (for "Music-on-Demand"), we developed a flexible protection scheme called MMP (for
"MultiMedia protection protocol") that effectively addresses this issue. Furthermore, MMP allows
to distribute real-time players "virtually free".
Audio servers may be used plainly for promotional purposes. E.g., museums may increase the
attractiveness of their WWW pages by adding some sound files, or mail-order services may add
sound excerpts to their server to increase their CD sales numbers. Opticom, a spin-off from
Fraunhofer, offers system solutions for this type of application. In spring 1996 (CeBit Hannover),
they successfully demonstrated an "audio-on-demand" application via T-Online together with the
Deutsche Telekom and a broadcasting station, the Suedwestfunk Baden-Baden.
Another music sales systems has been developed by Cerberus Sound & Vision. The company uses
a personalized real-time Layer-3 player and a proprietary encryption scheme to sell sound files via
the Internet on a "per song" base. Music servers and mirror sites are currently located in London,
New York, Tokyo and Rio; Melbourne and Berlin will follow soon.
"Audio-on-the-Internet" is currently a very popular topic. It does not only comprise audio file
transfers with download times as low as possible, but also streaming audio applications, like
"Internet Radio". As Layer-3 offers a sound quality "better than shortwave" at a bitrate of 16 kbps
(and, with some modifications, may even be useful at 8 kbps), various companies currently work on
this Internet subject - e.g., Opticom or Telos.
In a partnership with Apple, Telos introduced in September 96 the Audioactive technology to
support "Internet Radio" applications with a live audio input processed by a Layer-3 NetCoder
Hardware.
In December 96, Microsoft announced to support MPEG Layer-3 as part of their
NetShow multimedia server technology.
As first multimedia authoring tools, "Director Multimedia Studio 2" and "SoundEdit 16"
(from Macromedia) use Layer-3 to generate compressed sound files for the "shockwave" format.
Layer-3 encoders and decoders are not only available as studio equipment, but also as ISA-bus
PC boards from Dialog 4, along with application software, or as low-cost (decoder only) PC
boards from NSM; recording and playback tools are also available from Proton Data, along with a
special decoder module (called "CenLay3") that allows to playback Layer-3 files via the parallel
printer port. Proton Data has also developed a "cutting tool" that allows to manipulate audio data at
Layer-3 level.
In addition, a file-oriented Layer-3 encoder and decoder (called "L3ENC" and "L3DEC") is
available as shareware for various platforms. Registration is processed by Opticom. Please note
that even for registered users, the use of the shareware is limited to "personal edition" purposes.
Real-time Layer-3 players
WinPlay3
"WinPlay3" allows the decoding simply by software on any Pentium PC in real time. A 80486 class
CPU with a built-in floating-point-unit will also allow some limited operation. For the availability of
supported modes, please refer to the following performance matrix:
Pentium
486DX4-133
486DX2-66
486DX-50
486DX-33
MPEG-1 stereo
ok
ok
-
-
-
MPEG-1 downmix*
ok
ok
ok
-
-
MPEG-1 mono
ok
ok
ok
ok
-
MPEG-2 stereo
ok
ok
ok
ok
-
MPEG-2 downmix
ok
ok
ok
ok
ok
MPEG-2 mono
ok
ok
ok
ok
ok
*downmix: the original stereo signal will be played back as a mono signal
"MPEG-1" = "MPEG-1 Layer-3", i.e. sample rates 32, 44.1 or 48 kHz
"MPEG-2" = "MPEG-2 Layer-3", i.e. sample rates 16, 22.05 or 24 kHz
On a Pentium-90, WinPlay3 consumes less than 30 % of the CPU power to decode Layer-3
stereo @ 44.1 kHz, or around 5 % of the CPU power to decode Layer-3 mono @ 16 kHz.
At least, a 8-bit stereo sound card is required. For full quality audio, a 16-bit card is recommended.
The card�s MCI driver should support sampling frequencies from 8 kHz to 48 kHz.
A standard VGA graphics card is required.
As WinPlay3 buffers up to 4 seconds of sound data due to the limitations of the Microsoft
Windows multitasking architecture, around 1 MByte free physical memory must be available.
WinPlay3 runs with the following operating systems: Microsoft Windows 3.1/3.11 (in extended 386
mode), Windows 95 und Windows NT (long file names not yet supported).
WinPlay3 supports file play back of *.mp3 files and direct play from an URL via HTTP. WinPlay3
can simply be integrated as an helper application in common browsers, for example Netscape or
Mosaic.
WinPlay3 is available at http://www.iis.fhg.de/departs/amm/layer3/winplay3/. The unregistered
player is limited to a reproduction time of 20 sec, i.e. it will playback each plain Layer-3 file only for
this time. If you want to use your player without limitation, you have to register your player with
Opticom.
MMP
As many applications require a player that is "free" for the user, the latest versions of WinPlay3
(starting with version 2.0) also support the new "MMP" ("MultiMedia protection protocol") format.
MMP is a very flexible data format that may support the following functions:
"unlocking" of the 20 sec playback time limitation
"copyright protection" by applying encryption methods to (part of) the data
"title associated data" (e.g. ISRC code, user data)
"expiry date" to allow only a limited use
More detailed information is available at http://www.iis.fhg.de/departs/amm/layer3/mmp/.
In a typical "audio-on-demand" application, the content provider may "on-the-fly" convert its plain
Layer-3 data into MMP data, by using a "MMP tagger" software (available at Opticom). The client
may use its unregistered player to playback these files without limitation - the player is "virtually
free". The client need not pay fees - this issue now may be covered at the server side.
MPEG Layer 3 Player
For Mac OS users, a real-time player called "MPEG Layer 3 Player" with a similar look and feel
(and similar features) like "WinPlay3" will be released very soon. This new player will (finally!)
replace the much simpler (and somewhat buggy) pre-version 0.99 beta that has been available from
http://www.iis.fhg.de/departs/amm/layer3/macplay3/.
Layer-3 Sound on CD-ROMs
CD-ROMs (and hard disks) have become most popular to store "multimedia" data. Even with the
advent of the new DVD standard, memory capacity will remain a precious resource for many
applications. For uncompressed stereo signals from a CD, more than 10 MByte are necessary to
store one minute of music. Using Layer-3, less than 1 MByte is enough for the same playing time.
And significantly less memory is necessary, if some limitations in performance are acceptable. As
CD-ROM readers (and pretty soon, writers too) have already gained a significant market share,
typical applications focus today on storing compressed sound files on CD-ROMs, introducing more
or better sound tracks into the product. Real application examples are video games, music
catalogues or encyclopedias with sound excerpts (e.g., "MusicFinder" by Sygna), or talking books
for blind people.
Since fall 96, Bertelsmann is selling their new CD-ROM encyclopedia "Discovery 97"
providing information to around 100.000 key words, with rich multimedia information
(e.g. more than 2400 coloured photos and images, 41 interactive maps, more than 30
minutes of movie clips, 27 slide shows) including 150 minutes of sound tracks coded
with MPEG Layer-3.
Layer-3 Sound on Silicon
Up to now, solid-state memories (RAMs, Flash-ROMs) are only used as audio storage devices in
special (niche) applications, as the costs per byte are much higher than with other types of media
(magneto-optical disks or magnetic tapes). Speech announcement systems for mass transit vehicles
(e.g., busses, subways or trains) are an example for such special applications, as the rough
environment requires to use ROM based memories. Since 1993, Meister Electronic manufactures
speech announcement systems with Layer-3, significantly reducing the precious memory capacity
and, at the same time, significantly improving the sound quality (compared with their older 64 kbps
PCM "phone sound").
Today, PC-Cards with Flash-ROMs are available, offering a memory capacity up to 100 MByte
and more, but at prohibitive high costs for a consumer application. Here, further advances in
memory and card technology may trigger a new interesting market segment of
"audio-chip-card"-applications. At a press conference in August 95 in Munich, Siemens Germany
announced the advent of a new cost-effective ROM technology called the "ROS chip" (ROS =
Record-on-Silicon). The first generation of ROS chips will be in production in 1997, with a storage
capacity of 64 Mbit; a next generation with 256 Mbit as well as a one-time user programmable
version will follow. The ROS chips will be embedded in the new "MultiMedia-Card" from Siemens,
a cost-effective card media that will store data, text, graphics, images and sound. Siemens has
already demonstrated a battery-powered audio player using a prototype "Audio-Card" containing
sound tracks coded with MPEG-Layer-3.
General Questions and Answers
Q: O.K., Layer-3 is obviously a key to many applications. Where are its limitations?
A: Well, Layer-3 is a perceptual audio coding scheme, exploiting the properties of the
human ear, and trying to maintain the original sound quality as far as possible.
In contrast, a dedicated speech codec exploits the properties of the human vocal tract, trying
to maintain the intelligibility of the voice signals as far as possible. Advanced speech coding
schemes (e.g., CS-ACELP [LD-CELP] as standardised by ITU as G.723.1 [G.728])
achieve a useful voice reproduction at bitrates as low as 5.3 [16] kbps, with a codec delay
below 40 [1] ms. At such very low bitrates, they behave superior to Layer-3 for pure voice
signals, and they offer the low delay that is necessary for full- duplex voice communications.
In the framework of MPEG-4, scalable audio coding schemes are devised that combine
speech coding and perceptual audio coding.
Q: You mentioned the codec delay. May I have some figures?
A: Well, the standard gives some figures of the theoretical minimum delay:
Layer-1: 19 ms (<50 ms)
Layer-2: 35 ms (100 ms)
Layer-3: 59 ms (150 ms)
Practical values are significantly above that. As they depend on the implementation, precise
figures are hard to give. So the numbers in brackets are just rough thumb values - real
codecs may show even higher values. So yes, there are certain applications that may suffer
from such a delay (like feedback links for remote reporter units). For many other
applications (like the ones mentioned above), delay is of minor interest.
Overview about the ISO-MPEG Standard - or: What is MPEG all about?
Q: What is "MPEG"?
A: MPEG is the "Moving Picture Experts Group", working under the joint direction of the
International Standards Organization (ISO) and the International Electro-Technical
Commission (IEC). This group works on standards for the coding of moving pictures and
audio. MPEG has created its own homepage, providing information on the what, where,
when and how of the standards.
Q: What is MPEG-1, -2, and so on?
A: MPEG approaches the growing need for multimedia standards step-by-step. Today,
three main "steps" are defined (MPEG-1, MPEG-2, MPEG-4).
MPEG-1: "Coding of Moving Pictures and Associated Audio for Digital Storage
Media at up to about 1.5 Mbit/s"
MPEG-2: "Generic Coding of Moving Pictures and Associated Audio Information"
MPEG-3: originally planned mainly for HDTV applications; later on, it was merged
into MPEG-2
MPEG-4: "Coding of Audio-Visual Objects"
Q: Are MPEG-3 and Layer-3 the same thing?
A: No! Layer-3 is a powerful audio coding scheme which certainly is part of the MPEG
standard. Layer-3 is defined within the audio part of both existing international standards,
MPEG-1 and MPEG-2. So please do not mix audio layers and MPEG standards!
Q: What is the status of MPEG-1?
A: Work on MPEG-1 is finished. The first three parts are standardized since 1992.
MPEG-1 consists of five parts:
IS-11172-1 ("System") describes synchronization and multiplexing of video and audio
signals.
IS-11172-2 ("Video") describes compression of video signals, focussing on
progressive scan video (and mainly aiming at "Video-on-CD" applications).
IS-11172-3 ("Audio") describes a generic audio coding family, with three
hierarchically compatible members (called "Layer-1", "Layer-2" and "Layer-3").
IS-11172-4 ("Compliance Testing") describes procedures for determining the
characteristics of coded bitstreams and the decoding process and for testing
compliance with the requirements stated in the other parts.
DTR-11172-5 ("Software Simulation") is a technical report about a full software
implementation of the first three parts of MPEG-1.
Q: What is the status of MPEG-2?
A: MPEG-2 currently consists of nine parts. The first three parts are standardized since
1994, with some amendments included later on. Other parts are at different levels of
completion.
IS-13818-1 ("System") describes synchronization and multiplexing of video and audio
signals; it is also standardised by ITU-T as H.222.
IS-13818-2 ("Video") describes a generic video coding tool set, supporting interlaced
scan; it is also standardised by ITU-T as H.262.
IS-13818-3 ("Audio") describes a backward compatible extension of MPEG-1 for
multichannel audio coding ("surround sound", "multilingual sound") and a
non-backward compatible extension to lower sample rates, to support sound
applications with limited audio bandwidth requirements.
IS-13818-4 ("Conformance Testing") describes procedures for determining the
characteristics of coded bitstreams and the decoding process and for testing
compliance with the requirements stated in the other parts.
DTR-13818-5 ("Software Simulation") is a technical report about a full software
implementation of the first three parts of MPEG-2.
IS-13818-6 ("System Extensions - Digital Storage Media Command and Control
(DSM-CC))" describes a set of protocols for client-server applications
CD-13818-7 ("Audio, Non-Backwards-Compatible (NBC) - Coding") describes an
improved audio coding scheme for mono- and stereophonic signals as well as for
multichannel sound
13818-8 ("Video, extension to 10-bit input samples") has been withdrawn, due to
insufficient interest.
IS-13818-9 ("Real-Time Interface Specification for Low-Jitter Applications") defines
timing constraints on the real-time delivery of MPEG-2 transport bitstreams.
WD-13818-10 ("Conformance Extensions - DSM-CC") describes the addendum to
IS 13818-4 for DSM-CC
Q: "NBC audio"?" What is the motivation for this working group? What are the results?
A: Well, during the work for multichannel audio coding (IS-13818-3), it turned out that
backwards compatible (BC) schemes suffer from the matrixing process. Matrixing is
required to allow a MPEG-1 decoder to playback all surround channels via its two
stereophonic channels. Unfortunately, some of the introduced quantisation noise may become
audible after dematrixing. All in all, during an ISO listening test in spring 1994, BC
multichannel coding performed poorer, compared to non-ISO coding schemes (e.g.,
Dolby�s AC-3). So the NBC working group currently develops a new audio coding
scheme. NBC audio achieves a significant better performance, not only for multichannel
surround sound, but even for monophonic signals (here targeting "true transparency" at 64
kbps). In spring 1996, ISO performed a listening test for 5-channel surround sound, and
NBC audio using a total bit-rate of 320 kbps scored better than Layer-2 BC at a bit-rate of
640 kbps. NBC audio will also become one of the MPEG-4 audio coding algorithms.
Q: How do I get the MPEG documents?
A: Well, you may contact ISO, or you order it from your national standards body. E.g., in
Germany, please contact DIN.
Q: Is some public C source available?
A: Well, there is "public C source" available on various sites, e.g. at
ftp://ftp.fhg.de/pub/layer3/ or at
ftp://ftp.tnt.uni-hannover.de/pub/MPEG/audio/mpeg2/public_software/ . This code has been
written mainly for explanation purposes, so do not expect too much performance.
Some Basics about MPEG Audio - or: What about Layer-1, Layer-2,
Layer-3?
Q: Talking about MPEG audio, I always hear "Layer 1, 2 and 3". What does it mean?
A: MPEG describes the compression of audio signals using high performance perceptual
coding schemes. It specifies a family of three audio coding schemes, simply called Layer-1,
Layer-2, and Layer-3. From Layer-1 to Layer-3, encoder complexity and performance
(sound quality per bitrate) are increasing.
The three codecs are compatible in a hierarchical way, i.e. a Layer-N decoder may be able
to decode bitstream data encoded in Layer-N and all Layers below N (e.g., a Layer-3
decoder may accept Layer-1,-2,-3, whereas a Layer-2 decoder may accept only Layer-1
and -2.)
Q: So we have a family of three audio coding schemes. What does the MPEG standard
define, exactly?
A: For each Layer, the standard specifies the bitstream format and the decoder. To allow for
future improvements, it does not specify the encoder, but an informative chapter gives an
example for an encoder for each Layer.
Q: What have the three audio Layers in common?
A: All Layers 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.
The decoder is much less complex. Its only task is to synthesize an audio signal out of the
coded spectral components.
All Layers use the same analysis filterbank (polyphase with 32 subbands). Layer-3 adds a
MDCT transform to increase the frequency resolution.
All Layers use the same "header information" in their bitstream, to support the hierarchical
structure of the standard.
All Layers have a similar sensitivity to biterrors. They use a bitstream structure that contains
parts that are more sensitive to biterrors ("header", "bit allocation", "scalefactors", "side
information") and parts that are less sensitive ("data of spectral components").
All Layers support the insertion of programm-associated information ("ancillary data") into
their audio data bitstream.
All Layers may use 32, 44.1 or 48 kHz sampling frequency.
All Layers are allowed to work with similar bitrates:
Layer-1: from 32 kbps to 448 kbps
Layer-2: from 32 kbps to 384 kbps
Layer-3: from 32 kbps to 320 kbps
The last two statements refer to MPEG-1; with MPEG-2, there is an extension for the
sampling frequencies and bitrates (see below).
Q: What are the main differences between the three Layers, from a global view?
A: From Layer-1 to Layer-3, complexity increases (mainly true for the encoder), overall
codec delay increases, and performance increases (sound quality per bitrate).
Q: What are the main differences between MPEG-1 and MPEG-2 in the audio part?
A: 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.
Q: Is this all compatible to each other?
A: 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 "half" sample rates.
Q: You mean: compatible!? With all these extra audio channels? Please explain!
A: 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.
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?
A: 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.
Advanced Features of Layer-3 - or: Why does Layer-3 perform so well?
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?
A: 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
Q: I see. Sounds to me as if Layer-3 is something like a "Layer-2++". Now, tell me more
about sound quality. How do you assess that?
A: 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
Q: Listening tests are certainly an expensive task. Is there really no alternative?
A: 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.
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
Opticom), 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,
feel free to contact nmr@iis.fhg.de.
Q: O.K., back to these listening tests and the performance evaluation. Come on, tell me
some results.
A: Well, for more details you should study one of these AES papers or the MPEG
documents. For 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 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).
Q: Very interesting! Tell me more about this recommendation!
A: 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
�Լ����
--
* Origin: NJU PowerBBS <bbs.nju.edu.cn>
Technical Details
Message-ID:
<65xfGKN@bbs.nju.edu.cn>Path:
arc1-ca.pugleaf.net!unknown!not-for-mailShow Full Headers
Complete Article Headers
X-Google-Language: ENGLISH,ASCII X-Google-Thread: 1103fa,dc48dda7c5100291,start X-Google-Attributes: gid1103fa,public From: Xiao_Liu.pbbs@bbs.nju.edu.cn (Xiao Liu) Subject: about MP3 Date: 1997/12/09 Message-ID: <65xfGKN@bbs.nju.edu.cn> X-Deja-AN: 296480003 Organization: PowerBBS NJU Station Newsgroups: cn.bbs.comp.multimedia