Many listeners accept the MP3 bit rate of 128 kbit/s as near enough to compact disc quality for them. This provides a compression ratio of approximately 11:1, although listening tests show that with a bit of practice many listeners can reliably distinguish 128 kbit/s MP3s from CD originals (to some listeners, 128 kbit/s is unacceptably low quality).
Even though differences may be perceptible, this is acceptable for some listeners in some listening environments, such as in a noisy moving vehicle or at a party. A good demontration 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'.
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
These values are probably overly optimistic (they are likely to be influenced by public relations) because the quality depends not only on the encoding file format, but also on the quality of the psychoacoustic algorithms used by the encoder. Typical Layer 1 encoders use simple psychoacoustics which result in a higher needed bit rate for transparent encoding.
Layer 1 encoding at 384 kbit/s, even with these simple psychoacoustics, is better than Layer 2 at 192...256 kbit/s
Layer 3 encoding at 112...128 kbit/s is worse than Layer 2 at 192...256 kbit/s.
[The assumed bit rates are not equivalent in quality and the qualities are not necessarily optimal (it is generally agreed that 112 to 128 kbit/s Layer 3 is not excellent sound), and therefore this comparison is probably not reliable as an objective source.]
More realistic bit rates are:
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
Comparing a new file format is typically done by comparing a medium quality encoder of the old format and a highly tuned encoder of the new format.
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 are no longer widely used:
ISO dist10 reference code
Xing
BladeEnc
ACM Producer Pro.
The quality of MP3 files depend on the quality of the encoder and the difficulty of the signal being encoded.
Good encoders produce acceptable quality at 128 to 160 Kbit/s and near-transparency at 160 to 192 Kbit/s.
Low quality encoders may never reach transparency, not even at 320 Kbit/s.
So it is pointless 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 this is subjective. A given bit rate suffices for some listeners but not for others. 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.
Additionally, an important feature of MP3 is that it is lossy — meaning that it removes information from the input in order to reduce data storage (usually disk space) and bandwidth (often considered in terms of cost) requirements. As with most modern lossy encoders, MP3 algorithms work hard to ensure that the parts it removes cannot be detected by human listeners by modeling the general characteristics of human hearing (see also "noise masking"). The most notable benefit resulting from these efforts is the tremendous savings in disk space while maintaining a reasonable and acceptable (although detectable) loss in fidelity.
If your aim is to archive sound files with no loss of quality (or work on the sound files in a studio), then of interest is Lossless compression algorithms that are currently capable of compressing 16-bit PCM audio by 38 to 80% (partly depending upon the characteristics of the audio itself) while leaving the audio identical to the original:
Free Lossless Audio Codec (FLAC)
Monkey's Audio (APE)
Shorten (SHN)
Lossless Predictive Audio Compression (LPAC)
Wavpack (WV)
True Audio (TTA)
Apple Lossless
It is important to understand the difference between audio for further processing and audio use only for playback. Lossless formats are strongly preferred for material which will be edited, mixed, or otherwise processed because the perceptual assumptions made by lossy coders may not hold true after processing. Most people find lossy coders acceptable for audio which will only be played. Nevertheless, 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. Lossless formats will produce the best possible result, at the expense of a lower compression ratio.
If MP3 audio needs to be decoded then re-encoded (e.g., to be aired on radio later), cascading lossy compression stages can significantly reduce the quality of the end result. To prevent this, keep audio data in its original state if further operating on it is necessary. If any operation needs to be done on MP3 data, such as cutting or merging audio or lowering bit rates, it is preferable to use software that works directly with the encoded data (such as "mp3DirectCut" and "MP3Gain") to prevent extra decoding-encoding steps.
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