Free Pony Theory ~ MP3 (spec, strength, limitation)

Joined: 23 Sep 2006 Posts: 1088

A thread to discuss the most effective way to use mp3 standard for blogging.

Joined: 23 Sep 2006 Posts: 1088

http://en.wikipedia.org/wiki/MP3

Design limitations

There are several limitations inherent to the MP3 format that can not be overcome by any MP3 encoder.

Newer audio compression formats such as Vorbis and AAC no longer have these limitations.

In technical terms, MP3 is limited in the following ways:

* Bit rate is limited to a maximum of 320 kb/s (while some encoders can create higher bit rates, there is little-to-no support for these higher bit rate mp3s)
* Time resolution can be too low for highly transient signals, may cause some smearing of percussive sounds although this effect is to a great extent limited by the psychoacoustical properties of the Musicam polyphase filterbank (Layer II). Pre-echo is concealed due to the specific time-domain characteristics of the filter.
* Frequency resolution is limited by the small long block window size, decreasing coding efficiency
* No scale factor band for frequencies above 15.5/15.8 kHz
* Joint stereo is done on a frame-to-frame basis
* Encoder/decoder overall delay is not defined, which means lack of official provision for gapless playback. However, some encoders such as LAME can attach additional metadata that will allow players that are aware of it to deliver seamless playback.

Nevertheless, a well-tuned MP3 encoder can perform competitively even with these restrictions.

Joined: 23 Sep 2006 Posts: 1088

http://jthz.com/mp3/

VBR really is a no-brainer. Near the beginning and ending of a song (assuming it starts and ends softly), where the volume is lower, and the music is less "demanding" in terms of its encodability, it makes sense to drop the bit rate, simply because there's not much there to encode, and the wasted space is overkill. In the middle of the song, where it may be more complicated, the idea of giving the encoder the option of "bumping up" the rate on a frame-by-frame basis is great! You may end up with a file that's the same overall size as a 160kbit/s CBR, but that uses frames as low as 32 on the really dead parts, and as high as 320 on the really tough parts. The bitrate is dynamically adapting to keep the quality constant. To know that the whole file isn't bloated where it isn't necessary, is a real bonus. On top of that:
If you've ever listened to an MP3 of a complex song at a very low bitrate, you've probably heard the famous watery swishing sound, and artifacts in the treble or high end of the spectrum (such as cymbals), or the muffled tones at bass sounds. With Variable Bit Rate you can keep the song at a stable relative quality level by adjusting the compression rate according to the complexity of the encoded audio.

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http://www.mp3-converter.com/bitrates.htm

Selecting Bit Rates for Encoding

Bit rate refers to the speed at which a bit stream (compressed audio data in this case) will travel, or the amount of bits per second. With MP3, you usually refer to it in terms of kbps or how many thousands of bits per second. Just like the way film works with a higher number of frames producing a better picture, the more audio data that flows in a time period, the smoother and clearer the sound. "So what real difference is there in sound quality?" you might ask. Check out this MP3 bit rate page (courtesy of MP3-Tech.org, a good tech resource) where Gabriel Bouvigne ran a comparison of one track recorded at different bit rates and noted the difference. Many MP3 enthusiasts will convert to MP3 at 128 kbps (kilobytes per second) as a good medium in sound quality and file size. However, I recommend 160 kbps or VBR encoding, especially if you are a musician or if you have a sensitive ear to musical quality.

Joined: 23 Sep 2006 Posts: 1088

Usefull software for ripping...

EAC (plus you have to download LAME for the codec itself)

http://www.mp3-converter.com/exact_audio_copy.htm

Joined: 23 Sep 2006 Posts: 1088

This article discuss the do's and don't of audio ripping. (Thos I usually let the software do its thing, but it's nice to know what's behind it.)

http://findarticles.com/p/articles/mi_m0FXG/is_7_12/ai_55127379/pg_3

* Synchronization, Overlapped Reads, Jitter Correction: These terms are many times used interchangeably to describe a method of reading data from an audio disc that is meant to make the extraction as accurate as possible. When reading audio, the program will reread sectors to make sure that the program is extracting audio accurately. An extreme example is presented by Andre Wiethoff in his notes about his Exact Audio Copy program: "In secure mode, this program reads every audio sector at least twice. That is one of the reasons why the program is [so] slow. But by using this technique, it could detect any nonidentical sectors. If an error occurs (read or sync error), the program keeps on reading this sector, until eight of 16 retries are identical, or a selectable number of times these 16 retries are read. So, in [the ] worst case, bad sectors are read up to 82 times! But this will help the program to find the best result by comparing all of the retries. If it is not sure that the stream is correct (at least it can be said at approximately 99.5 percent), the program will tell the user where the (possible) read error occurred."

Overlapped and synchronized reading refers to a process of reading a certain amount of data, then going back and rereading a portion of that data and comparing it to the first read. If there is a mismatch, the data is read again from a different offset, then compared again until there is a match. This ensures that the proper data is being read even if the head positioning is not exact.

* Normalization: Different CDs are recorded at different volumes. When you make a compilation disc, there may be differences in the volume of the different songs, which can be very annoying. Normalization fixes this and sets all tracks to the same volume level.

* Track Offset: The track offset is the position of the read head in relation

to what the correct position should be. Usually the head will be a fixed offset before or after the correct read position. Once the offset is determined and adjusted for in the program, it should then position correctly on all CDs.

* Big-endian and Little-endian or Reverse Byte Order: Some audio extraction tools extract and store the files they extract in "big-endian" order, placing the "big end" at the lowest storage address used. Others store automatically in "little-endian" order, storing the "little end" byte at the lowest storage address. When tracks are extracted and stored in little-endian byte order, the files they create will sound like a bunch of static. Big-endian order stores the "big end" or most significant byte first at the lowest storage address. This approach must be reversed to be able to extract a track correctly.

* Swap Channels: Some drives swap the left and right channels when extracting audio. Some software allows you to correct this.

Joined: 23 Sep 2006 Posts: 1088

Comparatif des formats audio FLAC - APE - MP3 - MP3PRO - MPC - OGG - WMA

http://cf2mweb.phpnet.org/comparatif_audio/pg2.php

see also:
http://www.head-fi.org/forums/f46...ogg-wma-french-305987/index2.html
http://www.moteldemoka.com/2008/12/13/lirremediable/#comment-428377

Joined: 23 Sep 2006 Posts: 1088

MP3: The Definitive Guide

MP3: The Definitive Guide
O'Reilly Media | ISBN: 1565926617 | 400 pages | May 15, 2000 | PDF

MP3, standing for MPEG-1, Layer 3, is a codec for compressing the size of audio files for digital distribution. Much more than a definition, MP3 is nothing less than a cultural and economic revolution on the Internet. Every day, hundreds of thousands of MP3 music files are searched for, shared, recorded and listened to by computer and Internet users of all kinds. Either alone or collected into massive download sites, the MP3 revolution is seriously threatening the traditional ways people find, listen to and create music. MP3 players and encoders are available for all major computer platforms, including Windows, Macintosh, Linux and BeOS. MP3 hardware players include portable players you can take with you jogging or exercising from manufactures like Sony, Diamond and Philips to home stereo and car stereo players.
MP3: The Definitive Guide introduces the power-user to just about all aspects of MP3 technology. It delves into detail on obtaining, recording, and optimizing MP3 files using both commercial, and Open Source methods. Coverage is complete for four platforms: Windows, Macintosh, Linux and BeOS. In depth chapters describe all aspects of the MP3 experience from distributing, streaming, broadcasting, converting and playing to archiving your collection. Readers will learn how to test their equipment, optimize their encoding times, evaluate their playback options, control and organize a collection, even burn their own CD's or distribute their own music to a massive worldwide audience over the Internet. In addition, the author fills readers in on the complex legal issues surrounding MP3 files. Everything you need to know to enjoy MP3 today and tomorrow is contained in this single volume. MP3 is here to stay, and the applications for this versatile compression format are expanding exponentially along with its user base. MP3: The Definitive Guide should appeal to a broad audience of users, from the those just getting into this exciting new technology, to those who want to fully immerse themselves in the complexities and possibilities that MP3 presents.

http://rapidshare.com/files/20316610/1565926617.rar
http://depositfiles.com/files/683967

Joined: 23 Sep 2006 Posts: 1088

Fig.4 shows the spectrum of this demanding signal as preserved by lossless coding, in this case the popular FLAC codec (at its slowest "8" setting). To all intents and purposes, it is identical to the spectrum of the original CD. The lossless coding is indeed lossless, which I confirmed by turning the FLAC file back to WAV (LPCM) and doing a bit-for-bit comparison with the signal used to generate fig.3. The bits were the same—the music will also be the same!

Fig.4 Spectrum of 500Hz-spaced multitone signal at –10dBFS, FLAC encoding (linear frequency scale, 10dB/vertical div.).

Fig.7 Spectrum of 500Hz-spaced multitone signal at –10dBFS, MP3 encoding at 320kbps (linear frequency scale, 10dB/vertical div.).

Fig.7 shows the spectrum produced by the MP3 encoder running at 320kbps. (This is the format used by Deutsche Grammophon for its classical downloads.) Again, all the tones are reproduced correctly, and the noise has dropped by around 6dB or so at higher frequencies and up to 15dB at lower frequencies. But it is still not quite as low as AAC at 128kbps below 1kHz or so.

http://www.stereophile.com/features/308mp3cd/index1.html

Joined: 23 Sep 2006 Posts: 1088

Psychoacoustics: Facts and Models

Product Description
Psychoacoustics – Facts and Models offers a unique, comprehensive summary of information describing the processing of sound by the human hearing system. It includes quantitative relations between sound stimuli and auditory perception in terms of hearing sensations, for which quantitative models are given, as well as an unequalled collection of data on the human hearing system as a receiver of acoustic information. In addition, many examples of the practical application of the results of basic research in fields such as noise control, audiology, or sound quality engineering are detailed. The third edition includes an additional chapter on audio-visual interactions and applications, plus more on applications throughout. Acoustic demonstrations on a CD included with this edition further illustrate and amplify basic and applied psychoacoustic phenomena. Reviews of previous editions have characterized it as "an essential source of psychoacoustic knowledge," "a major landmark ," and a book that "without doubt will have a long-lasting effect on the standing and future evolution of this scientific domain."

About the Author

Hugo Fastl is Professor of Technical Acoustics in the Faculty of Electrical Engineering and Information Technology at the Technical University München, Germany. He graduated 1969 in Music from the Academy of Music München, and 1970, 1974, and 1981 he earned at the Technical University München the degrees of Dipl.-Ing., Dr.-Ing., and Dr.-Ing. habil., respectively. His research interests are basic psychoacoustics and its applications in fields like audio-communication, noise control, sound quality design, audiology, or music. In 1987 he was elected Guest Professor of Osaka University, Japan, and in 1990 he became a Fellow of the Acoustical Society of America. He is head of the committee "Auditory Acoustics" of the Society for Information Technology (ITG), and with the German Acoustical Society (DEGA) he is member of the Board of Directors, and Treasurer. In 1983 he won the Award of the Society for Information Technology (ITG), in 1991 the Research Award in Audiology of the Forschungsgemeinschaft Deutscher Hörgeräte-Akustiker, in 1998 the Research Award of the Japan Society for the Promotion of Science (JSPS), and in 2003 the Rayleigh Medal of the Institute of Acoustics (UK).

Eberhard Zwicker was one of the worlds top authorities in psychoacoustics. In his labs in Stuttgart and München he educated scientists and engineers who hold now key positions in basics and applications of (psycho-)acoustics. From his many honors, the Silver Medal of the Acoustical Society of America and the degree of Honorary Member of the Audio Engineering Society are among the more outstanding.

http://sharebee.com/ec7f8f69