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Inside the MP3 Codec
Waveforms
Breaking It Down
Lossy Compression
Masking Effects
Bitrates
Freedom of Implementation
Who Defines Imperceptible?
The Huffman Coding
MP3 Decoding
Anatomy of an MP3 File
ID3 Space
Frames Per Second
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Inside the MP3 Codec - Page 7
featured with permission from MP3: The Definitive Guide by Scot Hacker

Freedom of Implementation

Interestingly enough, the MP3 specification (ISO 11172-3) does not specify exactly how the encoding is to be accomplished. Rather, it outlines techniques and specifies a level of conformance; in other words, it tells developers that their resulting MP3 files must meet certain structural criteria.[10] This is necessary for the same reason that any standard exists: To allow for the proliferation of MP3 encoders and players by various vendors and developers. The specification only serves to guarantee a baseline consensus in the community regarding how certain things will operate. An encoder developed according to the MP3 specification will be capable of outputting a "compliant bitstream" that can be played successfully with any MP3-compliant decoder, just as you can create a JPEG file in any image editor under any operating system and expect it to display properly in any JPEG-compliant image viewer on any operating system.

It's important to maintain the distinction between the primary developers of the codec itself, The Fraunhofer Institute, and the committee that codified the work of Fraunhofer into the MPEG-I Layer 3 specification, the International Standards Organization (ISO). Standards are often created this way: A company produces a technology, other companies apply to become a part of the standards-creation process, and together they lay down the laws of implementation so that all vendors can compete around that technology. Note, however, that just because MP3 has been standardized by ISO does not mean that Fraunhofer (and their partners Thomson Multimedia) don't still hold the patent on the technology itself. As you'll see in Chapter 7, The Not-So-Fine-Print: Legal Bits and Pieces, Fraunhofer's patent is being aggressively exercised, making it difficult for small-time developers to affordably implement the ISO standard.

In any case, while the standard specifies exactly how decoding is to be accomplished, it only provides sample implementations (one simple and one complex) for encoding. As a result, there's a certain degree of headroom available for developers to make up some of the rules as they go along. In general, encoder developers work toward two goals: speed and quality. While there is some difference in the quality of audio files output by various encoders (as you'll see in Chapter 5), there are vast differences in the speed at which encoders operate. Sometimes, encoding speed comes at a distinct disadvantage to the quality of the resulting bitstream, though this is not necessarily the case.

A good example of the kind of freedom left to developers is the fact that the MP3 standard does not specify exactly how to treat the upper end of the spectrum, above 16kHz. Since human auditory perception begins to diminish greatly (with age and exposure to loud volumes) between 16kHz and 20kHz, some developers have historically chosen to simply chop off frequencies above 16kHz, which can be beneficial at low bitrates, since it leaves more bits available for encoding more audible frequencies. Xing, for example, did this with the first versions of their very fast codec. Later, they rewrote their codec to handle frequencies up to 20kHz (probably at the behest of the audiophile MP3 community).

 

NOTE

If you're curious about the upper and lower thresholds of your own hearing, download a sine wave generation program for your platform and run some tests. If you find a graphical program, you can simply turn the dial or drag the slider up the frequency spectrum until you can no longer hear it. If the program works from the command line, you can either generate sweep frequencies or generate a series of files at different frequencies at the upper end of the range and play them in sequence. BeOS and Linux users should check out a utility called sinus, while users of any platform can generate pure tones through any of the many simple synthesizer programs available at your favorite software library. The potential problem with running this kind of test lies in the fact that your playback hardware may itself not be capable of reproducing frequencies above, say, 17kHz. A test like this is best conducted on the highest quality equipment you can find.

 

Other Considerations

In addition to the general principles outlined in this chapter, the MP3 codec does a lot of additional work maintaining frequency tables, storing and allocating bits optimally, handling user options set at encode time, and the like. While we don't cover everything the encoder is responsible for exhaustively, here are a few of the more important additional chores the encoder must tackle.

Dipping into the reservoir

Because the bitrate is taken into consideration at every time frame, there will inevitably be certain frames of such complexity that they cannot be adequately coded to adhere to the limitations imposed by the chosen bitrate. In such a case, the MP3 spec allows for a "reservoir of bytes," which acts as a sort of overflow buffer when the desired amount of data cannot be stored in the given timeframe. In actual practice, this reservoir is not a separate storage space in the file, but rather the "empty space" left over in frames where the necessary information was encoded into the available space with room to spare. In other words, the byte reservoir is a portion of the algorithm designed to rob Peter and pay Paul.

While the CD and DAT audio formats typically offer 16 bits of resolution, the processing of a very complex musical passage may result in only four or six bits of resolution being encoded into the final bitstream,[11] since there isn't enough storage space allocated to handle the data needs of each frame. What can't be drawn from the reservoir will simply result in an audible degradation of the signal quality. Thus, the byte reservoir is only a partial solution to the loss of signal quality in complex passages. The only real solution to quality loss is to encode the signal at a higher bitrate.

The joint stereo effect

Most people have had an opportunity at some point to listen to a stereo system with a separate subwoofer attached (in fact, most better-quality computer speaker systems consist of two or four satellite speakers and a separate subwoofer). And as you may have noticed, the placement of satellite speakers is critical to high-quality audio reproduction, whereas the placement of the subwoofer is almost entirely irrelevant-people stuff subwoofers under desks, behind couches, or integrate them with other pieces of living room furniture. The reason it's possible to do this without affecting sound quality is because the human ear is largely insensitive to the location of the source of sounds at the very low and very high ends of the frequency spectrum.

The MP3 spec optionally exploits this aspect of human psychoacoustics as well. A file being encoded in stereo is by definition twice as large as a monophonic file. However, this file size doubling effect can be somewhat mitigated by combining high frequencies across the left and right tracks into a single track. This is done during the encoding phase by selecting the "joint stereo" option in the encoder's preferences, or by passing an appropriate command-line option to the encoder (there are actually several subtle differences between the various joint stereo encoding "modes"-more on that in Chapter 5). Since you might not be able to tell which speaker very high signals are emanating from anyway, there may be no point in storing that data twice.

 

NOTE

Some hard-core audiophile tweaks claim that bass sounds are not entirely nondirectional, only that they're less so than mid- and high-frequency sounds. Listeners with ears trained this well are probably not much interested in MP3 to begin with, but those listeners might be able to tell the difference in high-frequency spatialization when comparing MP3 to unencoded audio.

 

When the joint stereo option is enabled, a certain amount of "steering information" is added to the file so that these sounds can be placed spatially with some approximation of accuracy during playback. This becomes especially important at the upper edge of the bass spectrum, where the ear becomes more sensitive to the spatial location of bass signals. Joint stereo (in "Intensity" mode) really is a low-fi solution best reserved for situations where you need to keep file size at an absolute minimum.

 
WARNING

The joint stereo option can in some instances introduce audible compression artifacts which can't be removed by increasing the bitrate. The only way to find out whether this is a problem for you is to experiment. If you don't like the results, re-encode without joint stereo enabled. Remember: Your ears don't lie.

 


Side Information

If joint stereo is used in M/S (middle/side) mode, the left and right channels aren't encoded separately. Instead, a "middle" channel is encoded as the sum of the left and right channels, while a " side" channel is stored as the difference between the left and the right. During the decoding process, side information is read back out of the frame and applied to the bitstream so that the original signal can be reconstructed as accurately as possible. The side information is essentially a set of instructions on how the whole puzzle should be re-assembled on the other end.


Next:  Who Defines "Imperceptible"?

 

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