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Subject: MPEG-FAQ: multimedia compression [1/9]

This article was archived around: 9 Nov 1996 09:32:20 GMT

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Archive-name: mpeg-faq/part1 Last-modified: 1996/06/02 Version: v 4.1 96/06/02 Posting-Frequency: bimonthly
=========================================================================== ~Subject: SECTION 0. - INTRO ==================================================== THE MPEG-FAQ [Version 4.1 - 1. June 1996] ==================================================== PHADE Software Inh. Dipl-Inform. Frank Gadegast Leibnizstr. 30 10625 Berlin, GERMANY Fon/Fax ++ 49 30 3128103 E-mail phade@powerweb.de Web site http://www.powerweb.de/mpeg It's the eights publication of this file. Lots of information has been changed (which has surely brought errors with it, see Murphy's Law). This eights compilation is very different to the previous one, Version 4.0. First: The location of this file is: Text-Version : URL: ftp://ftp.powerweb.de/mpeg/faq/mpegfa41.zip [194.77.15.46] HTML-Version : URL: http://www.powerweb.de/mpeg/faq/ My MPEG-related software and my DOS-ports of several programs can be found there too. Second: "The Internet MPEG Audio Archive" is there ! Our brilliant collecting of everything that belongs to MPEG audio. For only DM 49,- ! Get it ! More than 400 MB of songs, documentation and utilities ! Read below, about how to Order ! Third: "The Internet MPEG CD-Rom" is still available ! The uniq collecting of everything that belongs to MPEG. For only DM 49,90 ! Get it ! More than 600 MB of movies, songs, documentation and utilities ! Read below, about how to Order ! Another CD-Rom containing material for MPEG-2 is about to get released ! It will be called the "MPEG-2 Movie Toolbox". Fourth: This FAQ has and the famous MPEG Archive has a complete new home now on the PowerWeb site ! The newest FAQ and other MPEG-related information and utilities for all platforms can always be loaded using WWW from: URL=http://www.powerweb.de/mpeg And surely, there are more interesting things to find ;o) I add my comments in brackets [], lines (---- or ====) seperate the chapters and questions. Please try and find out more information yourself. I had enough to do by getting and preparing this information. And only bother me with file- request if its not possible for you to get it somewhere else !!! If you want to contribute to this FAQ in any way, please email directly too (probably by replying to this posting): mpegfaq@powerweb.de If you want to contribute to the MPEG Archive, please upload via ftp to ftp://ftp.powerweb.de/incoming/mpeg and notity mpeg@powerweb.de via e-mail about your contribution. Other usefull information related to MPEG can be e-mailed to mpeg@powerweb.de Or send any additional information via fax or e-mail. Enjoy MPEG, KeyJ "MPEG" Phade (Frank Gadegast) ------------------------------------------------------------------------------- ~Subject: Disclaimer I HAVE NOTHING TO DO WITH THE NAMED COMPANIES, NO BUSINESS, IT'S JUST MY PERSONAL INTERESTED. COMPANIES ARE NAMED, BECAUSE THEY ARE THE FIRST, BRINGING REAL MULTIMEDIA TO THE WORLD. SURE I MAKE ADVERTS FOR THEM WITH THIS FAQ, BUT HOPE- FULLY YOU, AS A READER OF THIS FAQ, WILL FORCE THEM TO PRODUCE MORE AND BETTER PRODUCTS. MOST ADDITIONAL INFORMATION IS WRITTEN AS PERSONAL COMMENT, AND SHOULD NOT BE TAKEN AS PROOFEN FACTS. INFORMATION IS PRESENTED "AS IS", COULD BE OUT OF DATE AND CANNOT BE GARANTIED TO BE THE TRUTH. THIS INFOMATION COMES WITHOUT WARRANTY OF ANY KIND, INCLUDING WITHOUT LIMITATION OF WARRANTIES OF MERCHANTABILITY, FITNESS FOR PARTICULAR PURPOSE AND NON-INFRINGEMENT. UNDER NO CIRCUMSTANCES AND UNDER NO LEGAL THEORY, TORT, CONTRACT, OR OTHERWISE, SHALL THE AUTHOR BE LIABLE TO YOU OR ANY OTHER PERSON FOR ANY INDIRECT, SPECIAL, INCIDENTAL, OR CONSEQUENTIAL DAMAGES OF ANY CHARACTER INCLUDING, WITHOUT LIMITATION, DAMAGES FOR LOSS OF GOODWILL, WORK STOPPAGE, COMPUTER FAILURE OR MALFUNCTION, OR ANY AND ALL OTHER COMMERCIAL DAMAGES OR LOSSES. Frank Gadegast ------------------------------------------------------------------------------- ~Subject: Copyright information THIS COMPILATION OF INFORMATION IS COPYRIGHTED BY THE AUTHOR AND MAINTAINER, CURRENTLY FRANK GADEGAST. ANY NON-COMMERCIAL USE OF IT, OR PARTS OF IT IS ALLOWED, UNTIL THE USE OF IT IS REPORTED TO THE AUTHOR AND THE COMPILATION IS KEPT UNCHANGED. ADDITONAL, IF PARTS OF IT ARE USED, INFORMATION HAS TO BE ADDED WITH THAT PART, WHO THE AUTHOR OF THAT PARTS IS, THAT IT BELONGS TO THE COMPLETE COMPILATION AND WHERE TO FIND THE COMPLETE COMPILATION. COMMERCIAL USE CAN BE GRANTED IN SPECIAL CIRCUMSTANCES, FEEL FREE TO ASK AND SEND A DESCRIPTION OF THE INTENDED USE, TO RECEIVE A CERTIFICATION. ANY NON-REPORTED OR NON-CERTIFIED COMMERCIAL USE OF THIS COMPILATION IS A VIOLATION OF GERMAN COPYRIGHT LAW ! ANY RE-PUBLICATION OF THE INFORMATION IN THIS COMPILATION SHOULD BE REPORTED TO THE AUTHOR AND SHOULD BE QUOTED IN THE NEW PUBLICATION. ANY RE-DISTRIBUTION OF THE COMPLETE FILE ON NON-COMMERCIAL ARCHIVES, LIKE FTP- OR FAQ-MIRRORS IS ALLOWED. ------------------------------------------------------------------------------- ~Subject: Digest format It should be possible to read this FAQ with a threaded newsreader or emacs in FAQ-mode to enable you, to jump from one question to another, because this FAQ is organized as a digest. You can move to the next question with the digest commands in gnus, rn or other newsreaders, or with a regex search for ^~Subject or ^--. ------------------------------------------------------------------------------- ~Subject: Recommendations Well, to stop some of the most enoying question, from those that do not read this FAQ at all, I recommend the following player/decoder and encoder. Search the FAQ for these words and download them BEFORE e-mailing to me ! DOS: VMPEG, MAPLAYPC and CMPEG, ENC11BIN Windows: VMPEG, SoftPeg, COOL 1.5.3 and Maplay 1.2 for Win32 Unix: XMPLAY and VCR CD-I's and Video-CDs are currently only supported by VMPEG and SoftPeg ! ------------------------------------------------------------------------------- ~Subject: What questions are getting answered in this FAQ ? SECTION 0. - INTRO Disclaimer Copyright information Digest format What questions are getting answered in this FAQ ? SECTION 1. - WHAT IS MPEG-VIDEO/VIDEO What is MPEG ? What is MPEG-Audio then ? What is the Audio Layer 3 then ? What is MPEG-1+ ? What is MPEG-2 ? What happened at the MPEG - NY meeting ? What's about Video-CD and CD-I ? SECTION 2. - PROFESSIONAL SOFTWARE SUBSECTION - DOS MPEG Encoder by Xing SUBSECTION - WINDOWS MPEG ARCADETM XingSound XingCD SUBSECTION - UNIX Xing Distributed Media Architecture NVR Research Kit Demo of NVR Digital Media Development Kit How will I get the NVR-Software ? SECTION 3. - FREE AVAILABLE SOFTWARE SUBSECTION - DOS layr_100 mpeg2ppm vmpeg cmpeg dmpeg secmpeg mpegstat enc11dos pvrg MPEG SUBSECTION - Windows XingIt mpgaudio SUBSECTION - WINDOWS-NT mpeg2ply mpegplay SUBSECTION - OS/2 mp SUBSECTION - X-WINDOWS and UNIX Berkeley's MPEG Tools MPEG-1 Video Software Encoder MPEG Video Software Decoder MPEG Video Software Analyzer MPEG Blocks Analyzer MPEG Video Software Statistics Gatherer xmg mpegstat mplex xmplay xplayer xmpeg.tk mpeg2encode / mpeg2decode mpegaudio maplay Scanning MPEG's ... MPEG decoder... MPEGTool What is "SECMPEG" ? PVRG-MPEG Codec wdgt SUBSECTION - VMS vms MPEG SUBSECTION - MacIntosh Sparcle Qt2MPEG Audio on Macintosh ?! SUBSECTION - Atari SUBSECTION - Amiga MPEG2DCTV SUBSECTION - NeXT MPEG_Play.app mpegnext SUBSECTION - SGI SECTION 4. - MPEG-RELATED HARDWARE MPEG audio Layer-3 Video-Maker Some MPEG chips Optibase ReelMagic Cinerama XingIt!-card MPEG-decompression hardware list Amiga CD32 SECTION 5. - MAILBOX-ACCESS Genoabox Xing Technologies BBS and fax SECTION 6. - FTP-ACCESS FTP-ACCESS - Overview MPEG-2 validation bitstreams Audio streams and utils Accessing Aminet Where will I find test-material for MPEG-encoders ? SECTION 7. - WWW-ACCESS Where is the WWW-home of this FAQ ? An Interactive Explanation on the Web ? Where is the WWW-demo of "The Internet MPEG CD-Rom" ? Which archive is mostly related to MPEG-Audio ? What's with Bryan Woodworth ftp-area ? Rock'n'Roll stored in MPEG on the Web ? Where can I find space movies coded in MPEG ? Movies on Web-site Where can I find fractal movies coded in MPEG ? Is qt2mpeg on the Web ? What are other good URL's ? SECTION 8. - MAIL ORDER The Internet MPEG CD-Rom Conversion, WWW and CD-Rom production service How can I order information from C-CUBE ? SECTION 9. - ADDITIONAL INFORMATION What are the MPEG standard documents ? So, the Xing decoder is cheating, right ? What is Aware Inc. doing ? Will MPEG be included in QuickTime ? What's about MPEG-2 software ? What about good MPEG Hardware encoders (Optivision) ? What's about CD-I ? What is the PCMotion Player ? What is the MPEG-2 ISO number ? Some papers about MPEG-audio Where can I find more documents about what Berkeley is doing ? Is there a book about MPEG ? Who are CD-I producers ? Where can I get VideoCD and CD-I coding ? Where can I do MPEG encoding ? What the problem with all these file extensions for MPEG-files ? How can I do RTP encapsulation of MPEG1/MPEG2 ? Wo kann ich den MPEG-standard bestellen ? SECTION 10. - WHERE TO FIND MORE INFOS What newsgroups discuss MPEG ? How can 'archie' help me ? SECTION 11. - QUESTIONS =========================================================================== ~Subject: SECTION 1. - WHAT IS MPEG-VIDEO/VIDEO ------------------------------------------------------------------------------- ~Subject: What is MPEG ? From comp.compression Mon Oct 19 15:38:38 1992 Sender: news@chorus.chorus.fr Author: Mark Adler <madler@alumni.caltech.edu> [71] Introduction to MPEG (long) What is MPEG? Does it have anything to do with JPEG? Then what's JBIG and MHEG? What has MPEG accomplished? So how does MPEG I work? What about the audio compression? So how much does it compress? What's phase II? When will all this be finished? How do I join MPEG? How do I get the documents, like the MPEG I standard? [ There is no newer version of this part so far. Whoever wants to update ] [ this description, should do the job and send it over. ] Written by Mark Adler <madler@alumni.caltech.edu>. Q. What is MPEG? A. MPEG is a group of people that meet under ISO (the International Standards Organization) to generate standards for digital video (sequences of images in time) and audio compression. In particular, they define a compressed bit stream, which implicitly defines a decompressor. However, the compression algorithms are up to the individual manufacturers, and that is where proprietary advantage is obtained within the scope of a publicly available international standard. MPEG meets roughly four times a year for roughly a week each time. In between meetings, a great deal of work is done by the members, so it doesn't all happen at the meetings. The work is organized and planned at the meetings. Q. So what does MPEG stand for? A. Moving Pictures Experts Group. Q. Does it have anything to do with JPEG? A. Well, it sounds the same, and they are part of the same subcommittee of ISO along with JBIG and MHEG, and they usually meet at the same place at the same time. However, they are different sets of people with few or no common individual members, and they have different charters and requirements. JPEG is for still image compression. Q. Then what's JBIG and MHEG? A. Sorry I mentioned them. Ok, I'll simply say that JBIG is for binary image compression (like faxes), and MHEG is for multi-media data standards (like integrating stills, video, audio, text, etc.). For an introduction to JBIG, see question 74 below. Q. Ok, I'll stick to MPEG. What has MPEG accomplished? A. So far (as of January 1996), they have completed the "Standard of MPEG phase I, colloquially called MPEG I. This defines a bit stream for compressed video and audio optimized to fit into a bandwidth (data rate) of 1.5 Mbits/s. This rate is special because it is the data rate of (uncompressed) audio CD's and DAT's. The standard is in three parts, video, audio, and systems, where the last part gives the integration of the audio and video streams with the proper timestamping to allow synchronization of the two. They have also gotten well into MPEG phase II, whose task is to define a bitstream for video and audio coded at around 3 to 10 Mbits/s. Q. So how does MPEG I work? A. First off, it starts with a relatively low resolution video sequence (possibly decimated from the original) of about 352 by 240 frames by 30 frames/s (US--different numbers for Europe), but original high (CD) quality audio. The images are in color, but converted to YUV space, and the two chrominance channels (U and V) are decimated further to 176 by 120 pixels. It turns out that you can get away with a lot less resolution in those channels and not notice it, at least in "natural" (not computer generated) images. <IMG SRC="yuv411.gif"> <IMG SRC="yuv422.gif"> <IMG SRC="yuv444.gif"> The basic scheme is to predict motion from frame to frame in the temporal direction, and then to use DCT's (discrete cosine transforms) to organize the redundancy in the spatial directions. The DCT's are done on 8x8 blocks, and the motion prediction is done in the luminance (Y) channel on 16x16 blocks. In other words, given the 16x16 block in the current frame that you are trying to code, you look for a close match to that block in a previous or future frame (there are backward prediction modes where later frames are sent first to allow interpolating between frames). The DCT coefficients (of either the actual data, or the difference between this block and the close match) are "quantized", which means that you divide them by some value to drop bits off the bottom end. Hopefully, many of the coefficients will then end up being zero. The quantization can change for every "macroblock" (a macroblock is 16x16 of Y and the corresponding 8x8's in both U and V). The results of all of this, which include the DCT coefficients, the motion vectors, and the quantization parameters (and other stuff) is Huffman coded using fixed tables. The DCT coefficients have a special Huffman table that is "two-dimensional" in that one code specifies a run-length of zeros and the non-zero value that ended the run. Also, the motion vectors and the DC DCT components are DPCM (subtracted from the last one) coded. Q. So is each frame predicted from the last frame? A. No. The scheme is a little more complicated than that. There are three types of coded frames. There are "I" or intra frames. They are simply a frame coded as a still image, not using any past history. You have to start somewhere. Then there are "P" or predicted frames. They are predicted from the most recently reconstructed I or P frame. (I'm describing this from the point of view of the decompressor.) Each macroblock in a P frame can either come with a vector and difference DCT coefficients for a close match in the last I or P, or it can just be "intra" coded (like in the I frames) if there was no good match. Lastly, there are "B" or bidirectional frames. They are predicted from the closest two I or P frames, one in the past and one in the future. You search for matching blocks in those frames, and try three different things to see which works best. (Now I have the point of view of the compressor, just to confuse you.) You try using the forward vector, the backward vector, and you try averaging the two blocks from the future and past frames, and subtracting that from the block being coded. If none of those work well, you can intra- code the block. The sequence of decoded frames usually goes like: IBBPBBPBBPBBIBBPBBPB... Where there are 12 frames from I to I (for US and Japan anyway.) This is based on a random access requirement that you need a starting point at least once every 0.4 seconds or so. The ratio of P's to B's is based on experience. Of course, for the decoder to work, you have to send that first P *before* the first two B's, so the compressed data stream ends up looking like: 0xx312645... where those are frame numbers. xx might be nothing (if this is the true starting point), or it might be the B's of frames -2 and -1 if we're in the middle of the stream somewhere. You have to decode the I, then decode the P, keep both of those in memory, and then decode the two B's. You probably display the I while you're decoding the P, and display the B's as you're decoding them, and then display the P as you're decoding the next P, and so on. Q. You've got to be kidding. A. No, really! Q. Hmm. Where did they get 352x240? A. That derives from the CCIR-601 digital television standard which is used by professional digital video equipment. It is (in the US) 720 by 243 by 60 fields (not frames) per second, where the fields are interlaced when displayed. (It is important to note though that fields are actually acquired and displayed a 60th of a second apart.) The chrominance channels are 360 by 243 by 60 fields a second, again interlaced. This degree of chrominance decimation (2:1 in the horizontal direction) is called 4:2:2. The source input format for MPEG I, called SIF, is CCIR-601 decimated by 2:1 in the horizontal direction, 2:1 in the time direction, and an additional 2:1 in the chrominance vertical direction. And some lines are cut off to make sure things divide by 8 or 16 where needed. Q. What if I'm in Europe? A. For 50 Hz display standards (PAL, SECAM) change the number of lines in a field from 243 or 240 to 288, and change the display rate to 50 fields/s or 25 frames/s. Similarly, change the 120 lines in the decimated chrominance channels to 144 lines. Since 288*50 is exactly equal to 240*60, the two formats have the same source data rate. Q. You didn't mention anything about the audio compression. A. Oh, right. Well, I don't know as much about the audio compression. Basically they use very carefully developed psychoacoustic models derived from experiments with the best obtainable listeners to pick out pieces of the sound that you can't hear. There are what are called "masking" effects where, for example, a large component at one frequency will prevent you from hearing lower energy parts at nearby frequencies, where the relative energy vs. frequency that is masked is described by some empirical curve. There are similar temporal masking effects, as well as some more complicated interactions where a temporal effect can unmask a frequency, and vice-versa. The sound is broken up into spectral chunks with a hybrid scheme that combines sine transforms with subband transforms, and the psychoacoustic model written in terms of those chunks. Whatever can be removed or reduced in precision is, and the remainder is sent. It's a little more complicated than that, since the bits have to be allocated across the bands. And, of course, what is sent is entropy coded. Q. So how much does it compress? A. As I mentioned before, audio CD data rates are about 1.5 Mbits/s. You can compress the same stereo program down to 256 Kbits/s with no loss in discernable quality. (So they say. For the most part it's true, but every once in a while a weird thing might happen that you'll notice. However the effect is very small, and it takes a listener trained to notice these particular types of effects.) That's about 6:1 compression. So, a CD MPEG I stream would have about 1.25 MBits/s left for video. The number I usually see though is 1.15 MBits/s (maybe you need the rest for the system data stream). You can then calculate the video compression ratio from the numbers here to be about 26:1. If you step back and think about that, it's little short of a miracle. Of course, it's lossy compression, but it can be pretty hard sometimes to see the loss, if you're comparing the SIF original to the SIF decompressed. There is, however, a very noticeable loss if you're coming from CCIR-601 and have to decimate to SIF, but that's another matter. I'm not counting that in the 26:1. The standard also provides for other bit rates ranging from 32Kbits/s for a single channel, up to 448 Kbits/s for stereo. Q. What's phase II? A. As I said, there is a considerable loss of quality in going from CCIR-601 to SIF resolution. For entertainment video, it's simply not acceptable. You want to use more bits and code all or almost all the CCIR-601 data. From subjective testing at the Japan meeting in November 1991, it seems that 4 MBits/s can give very good quality compared to the original CCIR-601 material. The objective of phase II is to define a bit stream optimized for these resolutions and bit rates. Q. Why not just scale up what you're doing with MPEG I? A. The main difficulty is the interlacing. The simplest way to extend MPEG I to interlaced material is to put the fields together into frames (720x486x30/s). This results in bad motion artifacts that stem from the fact that moving objects are in different places in the two fields, and so don't line up in the frames. Compressing and decompressing without taking that into account somehow tends to muddle the objects in the two different fields. The other thing you might try is to code the even and odd field streams separately. This avoids the motion artifacts, but as you might imagine, doesn't get very good compression since you are not using the redundancy between the even and odd fields where there is not much motion (which is typically most of image). Or you can code it as a single stream of fields. Or you can interpolate lines. Or, etc. etc. There are many things you can try, and the point of MPEG II is to figure out what works well. MPEG II is not limited to consider only derivations of MPEG I. There were several non-MPEG I-like schemes in the competition in November, and some aspects of those algorithms may or may not make it into the final standard for entertainment video compression. Q. So what works? A. Basically, derivations of MPEG I worked quite well, with one that used wavelet subband coding instead of DCT's that also worked very well. Also among the worked-very-well's was a scheme that did not use B frames at all, just I and P's. All of them, except maybe one, did some sort of adaptive frame/field coding, where a decision is made on a macroblock basis as to whether to code that one as one frame macroblock or as two field macroblocks. Some other aspects are how to code I-frames--some suggest predicting the even field from the odd field. Or you can predict evens from evens and odds or odds from evens and odds or any field from any other field, etc. Q. So what works? A. Ok, we're not really sure what works best yet. The next step is to define a "test model" to start from, that incorporates most of the salient features of the worked-very-well proposals in a simple way. Then experiments will be done on that test model, making a mod at a time, and seeing what makes it better and what makes it worse. Example experiments are, B's or no B's, DCT vs. wavelets, various field prediction modes, etc. The requirements, such as implementation cost, quality, random access, etc. will all feed into this process as well. Q. When will all this be finished? A. I don't know. I'd have to hope in about a year or less. Q. How do I join MPEG? A. You don't join MPEG. You have to participate in ISO as part of a national delegation. How you get to be part of the national delegation is up to each nation. I only know the U.S., where you have to attend the corresponding ANSI meetings to be able to attend the ISO meetings. Your company or institution has to be willing to sink some bucks into travel since, naturally, these meetings are held all over the world. (For example, Paris, Santa Clara, Kurihama Japan, Singapore, Haifa Israel, Rio de Janeiro, London, etc.) Q. Well, then how do I get the documents, like the MPEG I standard ? A. MPEG is a ISO standard. It's exact name is ISO CD 11172. The standard consists of three parts: System, Video, and Audio. The System part (11172-1) deals with synchronization and multiplexing of audio-visual information, while the Video (11172-2) and Audio part (11172-3) address the video and the audio compression techniques respectively. You may order it from your national standards body (e.g. ANSI in the USA) or buy it from companies like OMNICOM phone +44 438 742424 FAX +44 438 740154 Or from 'ISO Online' at http://www.iso.ch/welcome.html ------------------------------------------------------------------------------- ~Subject: What is MPEG-Audio then ? From: "Harald Popp" <POPP@iis.fhg.de> From: mortenh@oslonett.no Date: Fri, 25 Mar 1994 19:09:06 +0100 Q. What is MPEG? A. MPEG is an ISO committee that proposes standards for compression of Audio and Video. MPEG deals with 3 issues: Video, Audio, and System (the combination of the two into one stream). You can find more info on the MPEG committee in other parts of this document. Q. I've heard about MPEG Video. So this is the same compression applied to audio? A. Definitely no. The eye and the ear... even if they are only a few centimeters apart, works very differently... The ear has a much higher dynamic range and resolution. It can pick out more details but it is "slower" than the eye. The MPEG committee chose to recommend 3 compression methods and named them Audio Layer-1, Layer-2, and Layer-3. Q. What does it mean exactly? A. MPEG-1, IS 11172-3, 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,-2,-3, with increasing encoder complexity and performance (sound quality per bitrate). The three codecs are compatible in a hierarchical way, i.e. a Layer-N decoder is 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 and -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. It does *not* specify the encoder to allow for future improvements, 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 or 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 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 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 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. Which Layer should I use for my application? A. Good Question. Of course, it depends on all your requirements. But as a first approach, you should consider the available bitrate of your application as the Layers have been designed to support certain areas of bitrates most efficiently, i.e. with a minimum drop of sound quality. Let us look a little closer at the strong domains of each Layer. Layer-1: Its ISO target bitrate is 192 kbps per audio channel. Layer-1 is a simplified version of Layer-2. It is most useful for bitrates around the "high" bitrates around or above 192 kbps. A version of Layer-1 is used as "PASC" with the DCC recorder. Layer-2: Its ISO target bitrate is 128 kbps per audio channel. Layer-2 is identical with MUSICAM. It has been designed as trade-off between sound quality per bitrate and encoder complexity. It is most useful for bitrates around the "medium" bitrates of 128 or even 96 kbps per audio channel. The DAB (EU 147) proponents have decided to use Layer-2 in the future Digital Audio Broadcasting network. Layer-3: Its ISO target bitrate is 64 kbps per audio channel. Layer-3 merges the best ideas of MUSICAM and ASPEC. It has been designed for best performance at "low" bitrates around 64 kbps or even below. The Layer-3 format specifies a set of advanced features that all address one goal: to preserve as much sound quality as possible even at rather low bitrates. Today, Layer-3 is already in use in various telecommunication networks (ISDN, satellite links, and so on) and speech announcement systems. Q. So how does MPEG audio work? A. Well, first you need to know how sound is stored in a computer. Sound is pressure differences in air. When picked up by a microphone and fed through an amplifier this becomes voltage levels. The voltage is sampled by the computer a number of times per second. For CD audio quality you need to sample 44100 times per second and each sample has a resolution of 16 bits. In stereo this gives you 1,4Mbit per second and you can probably see the need for compression. To compress audio MPEG tries to remove the irrelevant parts of the signal and the redundant parts of the signal. Parts of the sound that we do not hear can be thrown away. To do this MPEG Audio uses psychoacoustic principles. Q. Tell me more about sound quality. How good is MPEG audio compression? And how do you assess that? A. Today, there is no alternative to expensive listening tests. During the ISO-MPEG-1 process, 3 international listening tests have been performed, with a lot of trained listeners, supervised by Swedish Radio. They took place in 7.90, 3.91 and 11.91. Another international listening test was performed by CCIR, now ITU-R, in 92. All these tests used the "triple stimulus, hidden reference" method and the so-called CCIR impairment scale to assess the audio 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 With perceptual codecs (like MPEG audio), all traditional parameters (like SNR, THD+N, bandwidth) are especially useless. Fraunhofer-IIS (among others) works on objective quality assessment tools, like the NMR meter (Noise-to-Mask-Ratio), too. If you need more informations about NMR, please contact nmr@iis.fhg.de Q. Now that I know how to assess quality, come on, tell me the results of these tests. A. Well, for details you should study one of those AES papers listed below. One main result is that for low bitrates (60 or 64 kbps per channel, i.e. a compression ratio of around 12:1), Layer-2 scored between 2.1 and 2.6, whereas Layer-3 scored between 3.6 and 3.8. This is a significant increase in sound quality, indeed! Furthermore, the selection process for critical sound material showed that it was rather difficult to find worst-case material for Layer-3 whereas it was not so hard to find such items for Layer-2. For medium and high bitrates (120 kbps or more per channel), Layer-2 and Layer-3 scored rather similar, i.e. even trained listeners found it difficult to detect differences between original and reconstructed signal. Q. So how does MPEG achieve this compression ratio? A. Well, with audio you basically have two alternatives. Either you sample less often or you sample with less resolution (less than 16 bit per sample). If you want quality you can't do much with the sample frequency. Humans can hear sounds with frequencies from about 20Hz to 20kHz. According to the Nyquist theorem you must sample at least two times the highest frequency you want to reproduce. Allowing for imperfect filters, a 44,1kHz sampling rate is a fair minimum. So you either set out to prove the Nyquist theorem is wrong or go to work on reducing the resolution. The MPEG committee chose the latter. Now, the real reason for using 16 bits is to get a good signal-to-noise (s/n) ratio. The noise we're talking about here is quantization noise from the digitizing process. For each bit you add, you get 6dB better s/n. (To the ear, 6dBu corresponds to a doubling of the sound level.) CD-audio achieves about 90dB s/n. This matches the dynamic range of the ear fairly well. That is, you will not hear any noise coming from the system itself (well, there is still some people arguing about that, but lets not worry about them for the moment). So what happens when you sample to 8 bit resolution? You get a very noticeable noise floor in your recording. You can easily hear this in silent moments in the music or between words or sentences if your recording is a human voice. Waitaminnit. You don't notice any noise in loud passages, right? This is the masking effect and is the key to MPEG Audio coding. Stuff like the masking effect belongs to a science called psycho-acoustics that deals with the way the human brain perceives sound. And MPEG uses psychoacoustic principles when it does its thing. Q. Explain this masking effect. A. OK, say you have a strong tone with a frequency of 1000Hz. You also have a tone nearby of say 1100Hz. This second tone is 18 dB lower. You are not going to hear this second tone. It is completely masked by the first 1000Hz tone. As a matter of fact, any relatively weak sounds near a strong sound is masked. If you introduce another tone at 2000Hz also 18 dB below the first 1000Hz tone, you will hear this. You will have to turn down the 2000Hz tone to something like 45 dB below the 1000Hz tone before it will be masked by the first tone. So the further you get from a sound the less masking effect it has. The masking effect means that you can raise the noise floor around a strong sound because the noise will be masked anyway. And raising the noise floor is the same as using less bits and using less bits is the same as compression. Do you get it? Q. I don't get it. A. Well, let me try to explain how the MPEG Audio Layer-2 encoder goes about its thing. It divides the frequency spectrum (20Hz to 20kHz) into 32 subbands. Each subband holds a little slice of the audio spectrum. Say, in the upper region of subband 8, a 6500Hz tone with a level of 60dB is present. OK, the coder calculates the masking effect of this sound and finds that there is a masking threshold for the entire 8th subband (all sounds w. a frequency...) 35dB below this tone. The acceptable s/n ratio is thus 60 - 35 = 25 dB. The equals 4 bit resolution. In addition there are masking effects on band 9-13 and on band 5-7, the effect decreasing with the distance from band 8. In a real-life situation you have sounds in most bands and the masking effects are additive. In addition the coder considers the sensitivity of the ear for various frequencies. The ear is a lot less sensitive in the high and low frequencies. Peak sensivity is around 2 - 4kHz, the same region that the human voice occupies. The subbands should match the ear, that is each subband should consist of frequencies that have the same psychoacoustic properties. In MPEG Layer 2, each subband is 750Hz wide (with 48 kHz sampling frequency). It would have been better if the subbands were narrower in the low frequency range and wider in the high frequency range. That is the trade-off Layer-2 took in favour of a simpler approach. Layer-3 has a much higher frequency resolution (18 times more) - and that is one of the reasons why Layer-3 has a much better low bitrate performance than Layer-2. But there is more to it. I have explained concurrent masking, but the masking effect also occurs before and after a strong sound (pre- and postmasking). Q. Before? A. Yes, if there is a significant (30 - 40dB ) shift in level. The reason is believed to be that the brain needs some processing time. Premasking is only about 2 to 5 ms. The postmasking can be up till 100ms. Other bit-reduction techniques involve considering tonal and non-tonal components of the sound. For a stereo signal you may have a lot of redundancy between channels. All MPEG Layers may exploit these stereo effects by using a "joint- stereo" mode, with a most flexible approach for Layer-3. Furthermore, only Layer-3 further reduces the redundancy by applying huffmann coding. Q. What are the downside? A. The coder calculates masking effects by an iterative process until it runs out of time. It is up to the implementor to spend bits in the least obtrusive fashion. For Layer 2 and Layer 3, the encoder works on 24 ms of sound (with 1152 sample, and fs = 48 kHz) at a time. For some material, the time-window can be a problem. This is normally in a situation with transients where there are large differences in sound level over the 24 ms. The masking is calculated on the strongest sound and the weak parts will drown in quantization noise. This is perceived as a "noise- echo" by the ear. Layer 3 addresses this problem specifically by using a smaller analysis window (4 ms), if the encoder encounters an "attack" situation. Q. Tell me about the complexity. What are the hardware demands? A. Alright. First, we have to separate between decoder and encoder. Remember: the MPEG coding is done asymmetrical, with a much larger workload on the encoder than on the decoder. For a stereo decoder, variuos real-time implementations exist for Layer-2 and Layer-3. They are either based on single-DSP solutions or on dedicated MPEG audio decoder chips. So you need not worry about decoder complexity. For a stereo Layer-2-encoder, various DSP based solutions with one or more DSPs exist (with different quality, also). For a stereo Layer-3-encoder achieving ISO reference quality, the current real-time implementations use two DSP32C and two DSP56002. Q. How many audio channels? A. MPEG-1 allows for two audio channels. These can be either single (mono), dual (two mono channels), stereo or joint stereo (intensity stereo (Layer-2 and Layer-3) or m/s- stereo (Layer-3 only)). In normal (l/r) stereo one channel carries the left audio signal and one channel carries the right audio signal. In m/s stereo one channel carries the sum signal (l+r) and the other the difference (l-r) signal. In intensity stereo the high frequency part of the signal (above 2kHz) is combined. The stereo image is preserved but only the temporal envelope is transmitted. In addition MPEG allows for pre-emphasis, copyright marks and original/copy marks. MPEG-2 allows for several channels in the same stream. Q. What about the audio codec delay? 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) The practical values are significantly above that. As they depend on the implementation, exact figures are hard to give. So the figures in brackets are just rough thumb values. Yes, for some applications, a very short delay is of critical importance. E.g. in a feedback link, a reporter can only talk intelligibly if the overall delay is below around 10 ms. If broadcasters want to apply MPEG audio coding, they have to use "N-1" switches in the studio to overcome this problem (or appropriate echo-cancellers) - or they have to forget about MPEG at all. But with most applications, these figures are small enough to present no extra problem. At least, if one can accept a Layer- 2 delay, one can most likely also accept the higher Layer-3 delay. Q. OK, I am hooked on! Where can I find more technical informations about MPEG audio coding, especially about Layer- 3? A. Well, there is a variety of AES papers, e.g. K. Brandenburg, G. Stoll, ...: "The ISO/MPEG-Audio Codec: A Generic Standard for Coding of High Quality Digital Audio", 92nd AES, Vienna 1992, pp.3336 E. Eberlein, H. Popp, ...: "Layer-3, a Flexible Coding Standard", 94th AES, Berlin 93, pp.3493 K. Brandenburg, G. Zimmer, ...: "Variable Data-Rate Recording on a PC Using MPEG-Audio Layer-3", 95th AES, New York 93 B. Grill, J. Herre,... : "Improved MPEG-2 Audio Multi-Channel Encoding", 96th AES, Amsterdam 94 And for further informations, please contact layer3@iis.fhg.de Q. Where can I get more details about MPEG audio? A. Still more details? No shit. You can get the full ISO spec from Omnicom. The specs do a fairly good job of obscuring exactly how these things are supposed to work... Jokes aside, there are no description of the coder in the specs. The specs describes in great detail the bitstream and suggests psychoacoustic models. Originally written by Morten Hjerde <100034,663@compuserve.com>, modified and updated by Harald Popp (layer3@iis.fhg.de). Harald Popp Audio & Multimedia ("Music is the *BEST*" - F. Zappa) Fraunhofer-IIS-A, Weichselgarten 3, D-91058 Erlangen, Germany Phone: +49-9131-776-340 Fax: +49-9131-776-399 email: popp@iis.fhg.de ------------------------------------------------------------------------------- ~Subject: What is the Audio Layer 3 then ? Informations about MPEG Audio Layer-3 Version 1.51 - 1. 95 This text is organized as a kind of Mini-FAQ (Frequently Asked Questions). It covers several topics: 1. ISO-MPEG Standard 2. MPEG Audio Codec Family ("Layer 1, 2, 3") 3. Applications 4. Products 5. Support by Fraunhofer-IIS 6. Shareware Information For further comments and questions regarding Layer-3, please contact: - layer3@iis.fhg.de For further informations about MPEG, you may also like to contact: - phade@powerweb.de 1. ISO-MPEG Standard Q: What is MPEG, exactly? 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 associated audio. Q: What is the status of MPEG's work, then? What about MPEG-1, -2, and so on? A: MPEG approaches the growing need for multimedia standards step-by- step. Today, three "phases" are defined: MPEG-1:"Coding of Moving Pictures and Associated Audio for Digital Storage Media at up to about 1.5 MBit/s" Status: International Standard IS-11172, completed in 10.92 MPEG-2:"Generic Coding of Moving Pictures and Associated Audio" Status: International Standard IS-13818, completed in 11.94 MPEG-3: does no longer exist (has been merged into MPEG-2) MPEG-4: "Very Low Bitrate Audio-Visual Coding" Status: Call for Proposals first deadline 1. 10. 95 Q: MPEG-1 and MPEG-2 are ready-for-use. How do the standards look like? A: Both standards consist of 4 main parts. The structure is the same for MPEG-1 and MPEG-2. -1: System describes synchronization and multiplexing of video and audio -2: Video describes compression of video signals -3: Audio describes compression of audio signals -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. Q: How do I get the MPEG documents? A: You order it from your national standards body. E.g., in Germany, please contact: DIN-Beuth Verlag, Auslandsnormen Mrs. Niehoff, Burggrafenstr. 6, D-10772 Berlin, Germany Phone: +49-30-2601-2757, Fax: +49-30-2601-1231 2. MPEG Audio Codec Family ("Layer 1, 2, 3") Q: Talking about MPEG audio coding, I heard a lot about "Layer 1, 2 and 3". What does it mean, exactly? 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,-2,-3, with increasing encoder complexity and performance (sound quality per bitrate) from 1 to 3. The three codecs are compatible in a hierarchical way, i.e. a Layer-N decoder is 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 and -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 or 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: "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), "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: A lot of new stuff! 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, three terms can 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: I heard something about a new NBC mode for MPEG-2 audio? What does it mean? A: "NBC" stands for "non-backwards compatible". During the development of the backwards compatible MPEG-2 standard, the experts encountered some trouble with the compatibility matrix. The introduced quantisation noise may become audible after dematrixing. Although some clever strategies have been devised to overcome this problem, the question remained how much better a non-compatible multichannel codec might perform. So ISO-MPEG decided to address that issue in a "NBC" working group - among the proponents are AT&T, Dolby, Fraunhofer, IRT, Philips, and Sony. Their work will lead to an addendum to the MPEG-2 standard (13818-8). Q: O.K., that should do for a first overview. Are there some papers for a more detailed information? A: Sure! You'll find more technical informations about MPEG audio coding in a variety of AES papers (AES = Audio Engineering Society). The AES organizes two conventions per year, and perceptual audio coding has been a topic since the middle of the 80s. Some interesting papers might be: K. Brandenburg, G. 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 S. Church, B. 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 E. Eberlein, H. Popp, et al.: "Layer-3, a Flexible Coding Standard", 94th AES, Berlin Mar. 93, pp. 3493 B. Grill, J. Herre, et al.: "Improved MPEG-2 Audio Multi-Channel Encoding", 96th AES, Amsterdam Feb. 94, pp. 3865 J. Herre, K. Brandenburg, et al.: "Second Generation ISO/MPEG Audio Layer-3 Coding", 98th AES, Paris Feb. 95 F.-O. Witte, M. Dietz, et al.: "'Single Chip Implementation of an ISO/MPEG Layer-3 Decoder", 96th AES, Amsterdam Feb. 94, pp. 3805 For ordering informations, contact: AES 60 East 42nd Street, Suite 2520 New York, NY 10165-2520, USA phone: (212) 661-8528, fax: (212) 682-0477 Another interesting publication: the "Proceedings of the Sixth Tirrenia International Workshop on Digital Communications", Tirrenia Sep. 93, Elsevier Science B.V. Amsterdam 94 (ISBN 0 444 81580 5). An excellent tutorial about MPEG-2 has recently been published in a German technical journal (Fernseh- und Kino-Technik); part 4, by E. F. Schroeder and J. Spille, talks about the audio part (7/8 94, p. 364 ff). And for further informations, please feel free to contact layer3@iis.fhg.de. 3. Applications Q: O.K., let us concentrate on one or two audio channels. Which Layer shall I use for my application? A: Good Question. Of course, it depends on all your requirements. But as a first approach, you should consider the available bitrate of your application as the Layers have been designed to support certain areas of bitrates most effectively. Roughly, today you can achieve a data reduction of around 1:4 with Layer-1 (or 192 kbps per audio channel), 1:6..8 with Layer-2 (or 128..96 kbps per audio channel), and 1:10..12 with Layer-3, (or 64..56 kbps per audio channel), and still the reconstructed audio signal will maintain a "CD-like" sound quality. This may be used as a first "thumb rule" - let's talk about details later on. Q: Why does the performance increase with the number of the Layer? Why does the standard define a family of audio codecs instead of one single powerful algorithm? A: Well, the MPEG standard has forged together two main coding schemes that offered advantages either in complexity (MUSICAM) or in performance (ASPEC). Layer-2 is identical with the MUSICAM format. It has been designed as a trade-off between sound quality per bitrate and encoder complexity. So it is most useful for the "medium" range of bitrates (96..128 kbps per channel). For higher bitrates, even a simplified version, the Layer-1, performs well enough. Layer-1 has originally been developed for a target bitrate of 192 kbps per channel. It is used as "PASC" within the DCC recorder. For lower bitrates (64 kbps per channel or even less), the Layer-2 format suffers from its build-in limitations, and with decreasing bitrate, artefacts become audible more and more. Here is the strong domain of the most powerful MPEG audio format, Layer-3. It specifies a set of unique features that all address one goal: to preserve as much sound quality as possible even at very low bitrates. Q: Wait a second! I understand that Layer-3 has been an important asset to the MPEG-1 standard, to address the high-quality low bitrate applications. With the advent of the "low sample rate extension (LSF)" in MPEG-2, is it still necessary to rely on Layer-3 to achieve a high-quality sound at low bitrates? A: Yes, for sure! Please, don't mix up MPEG-1 and MPEG-2 LSF. MPEG-2 LSF is useful only for applications with limited bandwidth (11.25 kHz, at best). For applications with full bandwidth, MPEG-1 Layer-3 at 64 or 56 kbps per channel achieves the best sound quality of all ISO codecs. For applications with limited bandwidth, MPEG-2 LSF Layer-3 provides an excellent sound quality at 56 kbps for monophonic speech signals and still a good sound quality at only 64 kbps total bitrate for stereo music signals (with around 10 kHz bandwidth). The latest MPEG ISO listening test (in September 94 at NTT Japan, doc. MPEG 94/437) proved the superior performance of Layer-3 in MPEG-1 and MPEG-2 LSF. Q: 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: Is there really no alternative to listening tests? A: No, there is not. With perceptual codecs, all traditional "quality" parameters (like SNR, THD+N, bandwidth) are rather useless, as any codec may introduce noise and distortions as long as it does not affect the perceived sound quality. So, listening tests are necessary, and, if carefully prepared and performed, lead to rather reliable results. Nevertheless, Fraunhofer-IIS works on objective sound quality assessment tools, too. There is already a first product available, the NMR meter, a real-time DSP-based measurement tool that nicely supports the analysis of