The short answer: 16-bit 44.1 KHz PCM encoding, when properly sampled
and played back, is close enough to perfect reproduction for human
hearing in virtually all situations that it's unequivocally "good
The main caveats:
- The material must be recorded and reproduced with properly
engineered sampling and playback systems. While this is not
particularly difficult or expensive for a competent engineer using
modern technology, there are a number of mistakes the engineer
(both equipment designer and recording engineer) could make that
could be mitigated by a higher sampling rate and/or more bit depth.
- There are situations where 16-bit depth will have an audible noise
floor. These do not occur "naturally" and the vast majority of
listeners and even audio engineers would have neither the
inclination nor care to spend the money to produce an environment
where this would occur. (The noise floor is below audible in places
such as a soundproofed cinema in a quiet neighbourhood.)
- This applies to the storage format only: intermediate processing
uses appropriate higher bit depths and sampling rates as necessary.
As a simplistic example for bit depth, when mixing one would
normally mix multiple 16-bit input siganls to a 24-bit output
signal and then scale that output signal back to 16 bits. A
simplistic example for sampling rate is that one might sample at 8x
or more the 44.1 KHz final frequency in order to use analogue
filters that distort the signal less when filtering out signals
above the 22.05 KHz Nyquist frequency.
Now to the details.
A seemingly little known fact of digital sampling of analogue signals
is that, so long as the sampled signal has no frequency components
above the Nyquist frequency of 1/2 the sampling
rate, a properly reproduced playback of that sample will be an exact
copy of the analogue input waveform. All those stair steps you see in
pictures of sampling? They're nonsense; that's a made-up waveform that
cannot be generated by a proper reproduction system because such a
signal would have the "steps" removed by the output filter. I'm not
going to go into more details on this here, but if you are not
convinced or just want to learn more, see Monty Montgomery's "D/A and
A/D | Digital Show and Tell" in
video (also on
Note that other answers here get this wrong, and it does seem to be
very difficult to believe for some people. As this post
puts it quite eloquently:
The concept of the perfect measurement or of recreating a waveform
perfectly may seem like marketing hype. However, in this case it is
not. It is in fact the fundamental tenet of the Nyquist-Shannon
Sampling Theorem on which the very existence and invention of
digital audio is based. From WIKI: “In essence the theorem shows
that an analog signal that has been sampled can be perfectly
reconstructed from the samples”. I know there will be some who will
disagree with this idea, unfortunately, disagreement is NOT an
option. This theorem hasn't been invented to explain how digital
audio works, it's the other way around. Digital Audio was invented
from the theorem, if you don't believe the theorem then you can't
believe in digital audio either!
This tells us that in theory, with what we know about human hearing
limits and the noise floors of professionally-designed low-noise
listening environments (such as recording studio or good cinema), the
frequency response and noise floor of 44.1 KHz 16-bit digital audio
recordings will be essentially perfect. (There's a lot more detail on
this in 24/192 Music Downloads...and why they make no
sense. As an
interesting aside, it also mentions that providing wider spectra may
actually make things worse: playback of ultrasonic signals of any
significant amplitude into standard analogue audio amplifiers may well
create intermodulation distortion products in the audio frequencies.)
So the question now becomes, can we do the reproduction well enough in practice?
Well, the way to do this is to test it, of course.
These sorts of tests have been rife with major problems, some as bad as comparing different recordings of the "same" material, such as an SACD remaster of an album against its original master mix from the CD. Even very skeptical experts on testing can accept badly-advised shortcuts such as not double-blinding the test. And of course the listening environment has a massive and difficult-to-correct-for influence on the audio. Even just small movements of your head can result in massive spectrum changes due to comb filtering.
That said, amongst the enormous number of bad tests, a few good ones have been done and they have all invariably shown that nobody, not even professional recording engineers or people with "golden ears," can tell the difference between 44.1 KHz 16-bit and higher rate/depth source recordings.
The canonical paper on this dates from 2006 or so: Audibility of a CD-Standard A/D/A Loop Inserted into High-Resolution Audio Playback. The abstract:
Claims both published and anecdotal are regularly made for audibly
superior sound quality for two-channel audio encoded with longer word
lengths and/or at higher sampling rates than the 16-bit/44.1-kHz CD
standard. The authors report on a series of double-blind tests
comparing the analog output of high-resolution players playing
high-resolution recordings with the same signal passed through a
16-bit/44.1-kHz “bottleneck.” The tests were conducted for over a year
using different systems and a variety of subjects. The systems
included expensive professional monitors and one high-end system with
electrostatic loudspeakers and expensive components and cables. The
subjects included professional recording engineers, students in a
university recording program, and dedicated audiophiles. The test
results show that the CD-quality A/D/A loop was undetectable at
normal-to-loud listening levels, by any of the subjects, on any of the
playback systems. The noise of the CD-quality loop was audible only at
very elevated levels.
I'd like point out particularly section 4 of the paper because I think may give some insight into how this whole "high-definition" audio" mess happened:
Though our tests failed to substantiate the claimed advantages of
high-resolution encoding for two-channel audio, one trend became
obvious very quickly and held up throughout our testing: virtually
all of the SACD and DVD-A recordings sounded better than most CDs—
sometimes much better. Had we not “degraded” the sound to CD quality
and blind-tested for audible differences, we would have been tempted
to ascribe this sonic superiority to the recording processes used to
make them. Plausible reasons for the remarkable sound quality of
these recordings emerged in discussions with some of the engineers
currently working on such projects. This portion of the business is
a niche market in which the end users are preselected, both for
their aural acuity and for their willingness to buy expensive
equipment, set it up correctly, and listen carefully in a low-noise
environment. Partly because these recordings have not captured a
large portion of the consumer market for music, engineers and
producers are being given the freedom to produce recordings that
sound as good as they can make them, without having to compress or
equalize the signal to suit lesser systems and casual listening
conditions. These recordings seem to have been made with great care
and manifest affection, by engineers trying to please themselves and
their peers. They sound like it, label after label. High-resolution
audio discs do not have the overwhelming majority of the program
material crammed into the top 20 (or even 10) dB of the available
dynamic range, as so many CDs today do. Our test results indicate
that all of these recordings could be released on conventional CDs
with no audible difference. They would not, however, find such a
reliable conduit to the homes of those with the systems and
listening habits to appreciate them. The secret, for two-channel
recordings at least, seems to lie not in the high-bit recording but
in the high-bit market.
Here are my references and some more reading if you want to get more deeply into this.
- Audibility of a CD-Standard A/D/A Loop Inserted into
High-Resolution Audio Playback.
The best study I know of on this, though there are probably others.
- Paul D. Lehrman, The Emperor's New Sampling Rate,
Mix magazine. This is what led me to the article above, and it
serves as a higher-level summary, along with some further
- Monty Montgomery, "D/A and A/D | Digital Show and Tell"
video (also on
text form. If
you don't instinctively think "rubbish" when you see a stair-step
waveform associated with digital sampling, you really need to see
this. Even if you prefer reading things, the video is well worth
watching as the demonstrations of what's going on are very clear.
- Monty Montgomery, 24/192 Music Downloads...and why they make no
science behind hearing and why you can't hear "better" than 44.1
KHz/16-bit, and some information on sampling. Includes 16-bit WAV
files with 0 dB and -105 dB tones if you want to try to hear the
full dynamic range of 16-bits. Also a long list of what listening
tests may be testing instead of the source recording frequency and
- image-line.com, Audio Myths & DAW Wars.
A quick recapitulation of various things that usually cause changes
in audio quality outside of source rate/depth. Oriented towards
people who do music production.
- Ethan Winer, High Definition Audio Comparison. Do your own personal test of
"high-definition" vs. 44.1 KHz/16-bit!
- Ethan Winer, Ethan's Magazine Articles and Videos. Lots of other good
information on audio, listening tests, gear, and so on.