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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 8:1...6:1
Layer 3: 112...128 kbit/s, compression 12:1...10:1
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out-of-date technologies, or even buggy encoders for the old format.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 and
not always respecting the ISO/IEC standard.
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. Moreover, even with the
same encoder and resulting file size, a constant bitrate MP3 may sound
much worse than variable bitrate MP3.
It is important to note that quality of an audio signal is subjective.
Placebo effect is rampant, with many users claiming to require a
certain quality level for transparency. Many of these users fail an
A/B test and are unable to distinguish files of a lower bitrate. 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, Apple Lossless, TTA, FLAC,
Windows Media Audio 9 Lossless (wma) and Monkey's Audio (among
others). Lossless formats are strongly preferred for material that
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.
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.
