MP3

MP3 is a popular digital audio encoding and lossy compression format invented and standardised in 1991 by a team of engineers working in the framework of the ISO/IEC MPEG audio committee under the chairmanship of Professor Hans Musmann (University of Hannover - Germany). It was designed to greatly reduce the amount of data required to represent audio, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners. In popular usage, MP3 also refers to files of sound or music recordings stored in the MP3 format on computers.

Quality of MP3 audio

Because MP3 is a lossy format, it is able to provide a number of different options for its "bit rate" -- that is, the number of bits of encoded data that are used to represent each second of audio. Typically rates chosen are between 128 and 256 kilobit per second. By contrast, uncompressed compact disc audio has a bit rate of 1378 kbit/s or 1411 kbit/s.

Related Topics:
Lossy - Bits - Kilobit - Compact disc - Kbit/s

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

Looking at bit rate of CD audio from the sector perspective:

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

Bit rate of CD-DA (audio CD) = 2352 bytes/sector x 75 sectors/s = 176,400 bytes/s = 172.27 KB/s = 0.17 MB/s = more than 10 MB per minute.*

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

Looking at bit rate of CD audio from the sampling rate perspective:

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

CD audio is sampled using PCM to 16 bits per channel, with two channels, at 44.1 kHz. Therefore, bit rate = (44100 samples/channel)/s x 16 bit/sample x 2 channels = 1411200 bit/s = 176400 bytes/s = 172.27 KB/s = 0.17 MB/s = more than 10 MB per minute.*

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

*Actual bit rate is higher, because of EFM, CIRC, L2 ECC, and so on.

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

(For purposes of comparison, bit rate of data CD = 2048 bytes/sector x 75 sectors/s = exactly 150 KB/s = about 8.8 MB per minute.)

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

MP3 files encoded with a lower bit rate will generally play back at a lower quality. With too low a bit rate, "compression artifacts" (i.e., sounds that were not present in the original recording) may appear in the reproduction. A good demonstration of compression artifacts is provided by the sound of applause: it is hard to compress because it is random, therefore the failings of the encoder are more obvious, and are audible as ringing.

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

As well as the bit rate of the encoded file, the quality of MP3 files depend on the quality of the encoder and the difficulty of the signal being encoded. For average signals with good encoders, many listeners accept the MP3 bit rate of 128 kibit/s as near enough to compact disc quality for them, providing a compression ratio of approximately 11:1. However, listening tests show that with a bit of practice many listeners can reliably distinguish 128 kibit/s MP3s from CD originals; in many cases reaching the point where they consider the MP3 audio to be of unacceptably low quality. Yet other listeners, and the same listeners in other environments (such as in a noisy moving vehicle or at a party) will consider the quality acceptable.

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

Fraunhofer Gesellschaft (FhG) publish on their official webpage the following compression ratios and data rates for MPEG-1 Layer 1, 2 and 3, intended for comparison:

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

• Layer 1: 384 kbit/s, compression 4:1
• Layer 2: 192...256 kbit/s, compression 6:1...8:1
• Layer 3: 112...128 kbit/s, compression 10:1...12:1

The differences between the layers are caused by the different psychoacoustic models used by them; the Layer 1 algorithm is typically substantially simpler, therefore a higher bit rate is needed for transparent encoding. However, as different encoders use different models, it is difficult to draw absolute comparisons of this kind.

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

Many people consider these quoted rates as being heavily skewed in favour of Layer 2 and Layer 3 recordings. They would contend that more realistic rates would be as follows:

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

• Layer 1: excellent at 384 kbit/s
• Layer 2: excellent at 256...384 kbit/s, very good at 224...256 Kbit/s, good at 192...224 Kbit/s
• Layer 3: excellent at 224...320 Kbit/s, very good at 192...224 Kbit/s, good at 128...192 Kbit/s

When comparing compression schemes, it is important to use encoders that are of equivalent quality. Tests may be biased against older formats in favour of new ones by using older encoders based on out-of-date technologies, or even buggy encoders for the old format. Due to the fact that their lossy encoding loses information, MP3 algorithms work hard to ensure that the parts lost cannot be detected by human listeners by modeling the general characteristics of human hearing (e.g., due to noise masking). Different encoders may achieve this with varying degrees of success.

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

A few possible encoders:

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

• LAME first created by Mike Cheng in early 1998. It is (in contrast to others) a fully LGPL'd MP3 encoder, with excellent speed and quality, rivaling even MP3's technological successors.
• Fraunhofer Gesellschaft: Some encoders are good, some have bugs.

Many early encoders that are no longer widely used:

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

• ISO dist10 reference code
• Xing
• BladeEnc
• ACM Producer Pro.

Good encoders produce acceptable quality at 128 to 160 Kibit/s and near-transparency at 160 to 192 kbit/s, while low quality encoders may never reach transparency, not even at 320 kbit/s. It is therefore misleading to speak of 128 kbit/s or 192 kbit/s quality, except in the context of a particular encoder or of the best available encoders. A 128 kbit/s MP3 produced by a good encoder might sound better than a 192 kbit/s MP3 file produced by a bad encoder.

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

It is important to note that quality of an audio signal is subjective. A given bit rate suffices for some listeners but not for others. Individual acoustic perception may vary, so it is not evident that a certain psychoacoustic model can give satisfactory results for everyone. Merely changing the conditions of listening, such as the audio playing system or environment, can expose unwanted distortions caused by lossy compression. The numbers given above are rough guidelines that work for many people, but in the field of lossy audio compression the only true measure of the quality of a compression process is to listen to the results.

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

If your aim is to archive sound files with no loss of quality (or work on the sound files in a studio for example), then you should use Lossless compression algorithms, currently capable of compressing 16-bit PCM audio to 38% while leaving the audio identical to the original, such as Lossless Audio (LA) or Apple Lossless (among others). Lossless formats are strongly preferred for material which will be edited, mixed, or otherwise processed because the perceptual assumptions made by lossy encoders may not hold true after processing. The losses produced by multiple stages of coding may also compound each other, becoming more evident when the signal is reencoded after processing. Lossless formats produce the best possible result, at the expense of a lower compression ratio.

Related Topics:
Lossless compression - Apple Lossless

~ ~ ~ ~ ~ ~ ~ ~ ~ ~

Some simple editing operations, such as cutting sections of audio, may be performed directly on the encoded MP3 data without necessitating reencoding. For these operations, the concerns mentioned above are not necessarily relevant, as long as appropriate software (such as mp3DirectCut and MP3Gain) is used to prevent extra decoding-encoding steps.

~ ~ ~ ~ ~ ~ ~ ~ ~ ~