Talk:MPEG-1 Audio Layer II

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This is an interesting article, but there are few things I either don't understand or perhaps don't agree with.

The statement that sub-band coders work in the time domain, while MP3 is a transform coder and works in the frequency domain could be inaccurate. Surely both MP2 and MP3 are sub-band coders in that they divide the frequency space to be encoded into a number of bands. There are several ways of doing this. One way is to use a set of filters, and then it should certainly be possible to operate in the time domain as suggested. However the input would also have to be divided up into time slots, with a decision made as to which filter+time slot combination to encode. Essentially each time slot+filter combination has to be associated with a number of bits for encoding, which 0 bits implying that the combination is not encoded at all.

This processing is clearly not linear.

MP3 works in pretty much this way, and my understanding was that the reason that MP3 gave greater compression than MP2 is because MP3 uses temporal masking as well as frequency masking. With temporal masking low level signals both before and after a masking signal can be removed - apparently.

Exactly! This is also the reason many people in broadcasting (incorrectly) assume that MP2 is 'more stable' or 'more reliable' than MP3, while nothing could be further from the truth. No system in broadcast uses will ever stop and start all the time, which would be the ONLY argument in favor of MP2 (since the stream has less time-based corrections to recreate a decodec signal). I have just changed parts in the article that erroneously state that 'MP2 is the de facto standard' in broadcasting, which it isn't. I have just finetuned an Opticod 7600 myself, using L3 Joint Stereo for the STL in between radio-studio and transmitter end. Thoroughly comparing L2 with L3 at 256 kbit/s ends up in favor of L3, clearly, since it has much more room to either mask or pass through source quality. 80.126.64.97 (talk) 23:47, 13 December 2007 (UTC)[reply]
The occasional anecdote does not define a standard. And an STL is not a broadcast. When you are in control of both ends of the system, you can use whatever you want (within the bounds of the relevant regs). When broadcasting to the random public, you have no control over what equipment they are using; and receivers cannot be counted upon to decode Layer 3. Many do; but many don't. Layer 2 is the defacto standard by virtue of being the overwhelmingly dominant audio encoding used with MPEG 1 and 2 broadcasts. That is what "defacto standard" means. It does not mean that it's the only game in town.
—überRegenbogen (talk) 04:54, 20 December 2008 (UTC)[reply]

For implementing filters it is nowadays computationally effective to use a DFT algorithm, rather than implement a set of filters. I suspect that both MP2 and MP3 do this. Speech data compression used to be done by filter banks until eventually it became obvious that using DFT was feasible, and more effective. It is possible that some audio compression tools were being developed before the use of digital filtering became widespread and DFT could be done effectively in real time. For example early developments in DCC (Digital Cassette Tape) and MiniDisc could perhaps have been based on filters rather than DFT processing. Some very early systems may even have used analogue filters. This is perhaps of historical interest.

There is a statement that MP2 uses 32 channels, while MP3 uses 576. This may be true - I don't know. It does seem to me though that since MP3 is capable of encoding a wide range of sampling frequencies and quantisation levels, that it may not actually use a fixed set of channels. In other words, does MP3 always use 576 channels, or is the number of channels somehow dependent on the input signal. There might, for example, not be much point in using 576 channels for voice input sampled at 8kHz.

The observation that at high bit rates MP2 is somehow "more accurate" than MP3 is interesting, though it's unfortunate that many broadcasters in the UK use lower bit rates, with only Radio 3 consistenly broadcasting at 192kbps. This is just another example of a not so well known observation, which goes like this:

Codec A is better than Codec B at low frequency data rates. What can we say at higher data rates? Answer: NOT MUCH - indeed PERHAPS NOTHING AT ALL

Some software purveyors have tried to convince us that a particular codec may be better by demonstrating that it does better at a given data rate. It does not necessarily follow that at different data rates the same properties will hold. For example, Microsoft's WMA codec is clearly better (I really believe this ...) than MP3 at rates of 48kbps and lower. This says absolutely nothing about their relative quality at rates of 96kbps and above, and compatibility factors would tend to favour MP3. This is because different algorithmic features may kick in at different compression rates. aacPlus and mp3Pro both use SBR (Spectral Band Replication) at low data rates, which enhances them relative to AAC and MP3, but at higher bit rates they probably just converge to plain AAC and MP3.

David Martland 07:51, 27 December 2005 (UTC)[reply]

I just had a look through this article and have a few comments:
Both MP2 and MP3 split the audio into different frequency sub-bands. After this MP3 uses a MDCT to resolve each sub-band into frequency components. Depending on the block length of the MDCT, this can either result in 192 or 576 frequency components. See http://wiki.hydrogenaudio.org/index.php?title=MP3
So, in response to above, both MP2 and MP3 do use filters, but only MP3 uses the MDCT. See also: http://www.cs.columbia.edu/~coms6181/slides/6R/mpegaud.pdf
The comments about Ogg Vorbis do not have citations and are too vague. There is no mention of what artifacts the glockenspiels caused or why Vorbis would avoid them. Vorbis still employs the MDCT, making it similar to many other codecs. Vorbis has also had its issues with artifacts, namely pre-echo, although these were really only a problem with the older encoders.
The explanation of why MP3 provides better compression than MP2 is just plain wrong. Again, please see the above pdf for a description of the design improvements of MP3 over MP2.

""incorrectly" called Musicam"[edit]

Musicam is the name used for MP2 in the specifications for DAB and Astra Digital Radio as well as in the BBCs DAB documents - it may not be a common name these days for MPEG 1 Layer 2 audio compression, but it seems to be common enough in older documents, leading me to think its not "incorrectly" called Musicam. --Kiand 02:18, 3 February 2006 (UTC)[reply]

Support. I deleted the corresponding passage. --Abdull 17:55, 7 March 2006 (UTC)[reply]

Links to MP3[edit]

Sorry if this has previously been mentioned, but why is history on MP2 stored in the MP3 article. (Just look at the top of this article and there is a link saying For details and a short historic introduction to MP2. Surely because it is related to MP2 then it should be in this article and not in the MP3 article. Matthuxtable 16:29, 15 March 2006 (UTC)[reply]

P.s. There's a section on my talk page if you want me to see your reply as I may forget to check back here. See User talk Matthuxtable Special Reply Section. Many thanks Matthuxtable 16:30, 15 March 2006 (UTC)[reply]

How about licensing and patent issues? Are they the same as for MP3?

I heard a rumor that the patents for MP2 are expired. True? The "For details and a short historic introduction to MP2, see MP3." is quite bad. — Omegatron 16:29, 20 July 2006 (UTC)[reply]

Too technical[edit]

The second section is just swirling with stuff that just goes right over my head and I consider myself pretty tech-savvy. I think it ought to either be removed or heavily rewritten to follow the KISS principle. — User:ACupOfCoffee@ 17:28, 3 October 2006 (UTC)[reply]

I added links for sampling rate and bitrate to the second section. Does that help? Daniel.Cardenas 20:10, 3 October 2006 (UTC)[reply]

Is this over-simplified?[edit]

This is the fundamental difference between Musicam (the MPEG1 Audio family) and the subsequent audio compression codecs. Musicam based codecs (MP2, MP3) have put first a focus on time domain critical audio sequences which are more typical in classical music and professional applications (studio) whereas subsequent codecs have ignored this demanding requirement to focus on less critical (frequency domain)light music audio materials.

The above text implies that there is a clear-cut class distinction between "time-domain-critical" music and "light music". It seems to say that codecs after Musicam (MP2, MP3) ignore time-domain fidelity. What is meant by "put a focus on" will I think be unclear to general readers.
I would prefer a clearer text that makes these points:
  • All of these codecs are lossy meaning that in order to compress data rate they introduce distortions in both the time and frequency domains
  • Successive codecs in the family achieve compression by increasingly complex methods that exploit the characteristic of human hearing that some distortions go unnoticed.
  • Time-domain distortion alone has been found to be almost imperceptible except for very critical listeners to some audio material. Cuddlyable3 12:41, 11 May 2007 (UTC)[reply]

FUD?[edit]

  • AAC and to a lesser extent Ogg Vorbis and AC-3 audio codecs are still affected by the same fundamental problem in the codec model that the triangle, kabuki, glockenspiel and crysaglott revealed -- coding signals with complex impulses and high energy transients are always poorly reproduced. //This is the fundamental difference between Musicam (the MPEG1 Audio family) and the subsequent audio compression codecs. Musicam based codecs (MP2, MP3) have put first a focus on time domain critical audio sequences which are more typical in classical music and professional applications (studio) whereas subsequent codecs have ignored this demanding requirement to focus on less critical (frequency domain)light music audio materials.

To me, the above looks like a FUD (Fear, Uncertainty, and Doubt) statement. Could someone please find a citation or remove it. The term "subsequent codecs" especially is far too general. I don't know much about AC-3 but I do know Ogg Vorbis allows frame sizes to be specified in the header to enable better handling of high energy transients. Additionally, a lot of tuning to handle high energy transients has been done because of previous Vorbis problems with pre-echo.

Please sign your posts. I think comparative information about rival codecs would be handled better on another page such as audio_compression or list_of_codecs than here. However it is unhelpful to bring amateur psychobabble jargon to this discussion. Cuddlyable3 08:31, 15 June 2007 (UTC)[reply]
None of that was intended to be 'psychobabble jargon'. MP2 uses frame sizes of 1152 samples to encode sound. Ogg Vorbis allows specification of 2 frame sizes at the start of a file from 64 - 8192 samples. Smaller frame sizes allow high-energy transients to be handled better. I'm not trying to handle a discussion of codecs or make a really confusing argument, I'm just pointing out why I think the statement is FUD and why it would just nice to have a citation for it. -Francis

In response to FUD concerns - the statement "Some important (mostly undocumented) events in the development of MP2 stand out." really started alarm bells ringing. If theses import developments are undocumented then the subsequent content is at best original research, although it seems to me a conspiracy theory about how the codec developers want to ruin music with newer codecs. I've decided to delete some of the most confusing and dubious "undocumented evens", if anyone can find some citations then perhaps they could be reinstated. JohnPW (talk) 18:24, 16 January 2009 (UTC)[reply]

Patent Expiration?[edit]

Does anyone know when the last patents expire for MPEG-1 Audio Layer II? US 4972484  which expires on July 21, 2008 seems to apply to MP2. Are there any other MP3 patents that apply to MP2? Jrincayc (talk) 15:15, 1 June 2008 (UTC)[reply]

User:Rcooley suggested I take a look at the article: "Low bit-rate coding of high-quality audio signals. An introduction to the MASCAM system" by G. Thiele, G. Stoll and M. Link, published in EBU Technical Review, no. 230, pp. 158-181, August 1988. This article includes features like scale factors, subband coding, and allocating different numbers of bits to different subbands for each sample that are later used in MPEG-1 Audio Layer II (MP2). I am trying to track down more information on the Pseudo-Quadrature Mirror Filters that is used for encoding and decoding the audio in MP2 since that is the remaining piece of MP2 that might be original enough to have a strong patent. Anybody have any information about the filters used in MP2? Jrincayc (talk) 00:48, 26 December 2008 (UTC)[reply]

Howto?[edit]

Howto was added to the "How the MP2 codec works". After reading over the section, I disagree that the section was providing instructions on how to do something. Can you be more specific? Jrincayc (talk) 12:21, 14 July 2008 (UTC)[reply]

Title[edit]

why is this article called "MPEG-1 Audio Layer II" but the mp3 one called "mp3" shouldn't they both by either "MPEG-1 Audio Layer 2" and "MPEG-1 Audio Layer 3" respectivly or simply "mp2" and "mp3"? -- Sonarpulse | Talk 05:27, 25 August 2008 (UTC)[reply]

mp2 is not as popular as mp3 so mp3 gets the popular name and mp2 gets the more technical name. Daniel.Cardenas (talk) 17:14, 25 August 2008 (UTC)[reply]

History of Development[edit]

I have deleted the unsourced paragraphs from the History of Development from MP2 to MP3 section, based on what seemed from the page history to be consensus by several wikipedia users and only opposed by one, who has now left the project. The statements were unsourced, and is so at variance with reality (if newer codecs are inferior, why are they being used, even replacing older ones?) that I doubt they could be true. Wingedsubmariner (talk) 15:11, 20 March 2009 (UTC)[reply]

Incorrect redirection from ISO/IEC 13818-3 to MPEG-1 Layer II[edit]

As far as I know the ISO/IEC 13818-3 (MPEG-2) extended all MPEG-1 Audio Layers, not only Layer II. See article MP3. The MP3 (MPEG-1 Layer III) was extended also in ISO/IEC 13818-3. Source: http://www.mpgedit.org/mpgedit/mpeg_format/mpeghdr.htm I suggest to remove redirection from ISO/IEC 13818-3 to MPEG-1 Layer II. There could be a separate article about ISO/IEC 13818-3 with informations about defined changes to MPEG-1 audio. --89.173.68.106 (talk) 11:13, 21 June 2009 (UTC)[reply]

As I see, there is also a wrong redirection from ISO/IEC 11172-3 to MPEG-1 Layer II. This ISO standard defined MPEG-1 audio with layers I, II and III. I suggest removal of this redirection. ISO/IEC 11172-3 could be a separate article for example with name MPEG-1 Audio containig short informations about Layers I, II and III. --89.173.68.106 (talk) 11:26, 21 June 2009 (UTC)[reply]

Encoders[edit]

Maybe something about software and hardware encoders? I think some are available e.g. for television broadcasters. 83.26.110.196 (talk) 18:38, 29 November 2011 (UTC)[reply]

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Patent[edit]

A good improvement to this article would be including information about the patent status (if all expired or not). I could not easily find these data, though... By the way, kuro5hin is dead, so the link "Patent Status of MPEG-1, H.261 and MPEG-2" is broken!--200.223.199.146 (talk) 11:46, 9 December 2016 (UTC)[reply]

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Rename to avoid confusion with specific versions[edit]

This article should be renamed to MP2 (audio coding format) to avoid confusion between the name that includes all versions (MP2) and names for specific versions (MPEG-1 Audio Layer II, MPEG-2 Audio Layer II). --Fernando Trebien (talk) 21:28, 11 March 2020 (UTC)[reply]

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