Audio quality is an interesting term, as it can be both a
fact and a subjective opinion. CDs are very much still in mass production today, but with the advent of commercial goliath: Apple’s iTunes, and growing rivals such as Spotify, hugely compressed .mp3
files have in many ways become the norm. Many people, and particularly the
youth of today have grown to accept this as the best solution for piping music
in to their ears, but the question is: are they right?
The recording and distribution of music has been through
many different stages during the past half century: evolving from analogue
recording methods like vinyl and cassette tape to digital CDs, DVDs and now
Bluray.
“An analogue recording is one where a property or characteristic of a
physical recording medium is made to vary in a manner analogous to the
variations in air pressure of the original sound.” - Wikipedia
“A digital recording is produced by converting the physical properties
of the original sound into a sequence of numbers, which can then be stored and
read back for reproduction.” - Wikipedia
Both analogue and digital systems have limitations, and there is debate
as to which is the best solution, but this a very extensive, ever on-going and
open ended discussion that I don’t want to get in to with this article. The
main point is that both of these methods are used in order to capture music in
as a raw a form as possible. If the resulting product is then presented on a
good enough sound system, it should be a near perfect reproduction of said
original sounds.
The focus of this article will be digital media, and as such a breakdown
of the key steps in producing such a recording is included below (courtesy of
Wikipedia):
Recording
· The analogue signal is transmitted from the
input device to an analogue-to-digital converter (ADC).
· The ADC converts this signal by repeatedly
measuring the momentary level of the analogue (audio) wave and then assigning a
binary number with a given quantity of bits (word length) to each measuring
point.
· The frequency at which the ADC measures the
level of the analogue wave is called the sample rate or sampling rate.
· A digital audio sample with a given word
length represents the audio level at one moment.
· The longer the word length the more exact is
the representation of the original audio wave.
· The higher the sampling rate the higher the
upper cut-off frequency of the digitised audio signal.
· The ADC outputs a sequence of samples that
make up a continuous stream of 0s and 1s.
· These numbers are stored onto recording media
such as magnetic tape, hard drive, optical drive or solid state memory.
Playback
· The sequence of numbers is transmitted from
storage into a digital-to-analogue converter (DAC), which converts the numbers
back to an analogue signal by sticking together the level information stored in
each digital sample, thus rebuilding the original analogue wave form.
·This signal is amplified and transmitted to
the loudspeakers.
From above, we can see that there are two main factors
affecting the quality of a digital recording:
Word Size
The number of bits used to represent a single audio wave (the word
size) directly affects the achievable noise level of a signal recorded with
added dither, or the distortion of an undithered signal. Increasing a sample's
word length by one bit doubles its possible values, likewise increasing the
potential accuracy of each sample and the fidelity of the recording to the
original. 24-bit recording is generally considered a current practical limit as
this word length allows a signal-to-noise ratio exceeding that of most analogue
circuitry, which by necessity must be used in at least two points in the
recording/playback chain
Sample
rate
The sample rate is even more important a consideration than the
word size. If the sample rate is too low, the sampled signal cannot be
reconstructed to the original sound signal. Hence the output will be different
from the input. The process of under sampling results in aliasing whereby the
high frequency components of the sound wave are represented as being lower than
they should be. This causes the output wave shape to be severely altered.
To overcome aliasing, the sound signal (or other signal) must be
sampled at a rate at least twice that of the highest frequency component in the
signal. This is known as the Nyquist-Shannon sampling theorem.
For recording music-quality audio the following PCM sampling rates
are the most common:
44.1 kHz 48 kHz 88.2 kHz 96 kHz 176.4 kHz 192 kHz
When making a recording, experienced audio recording and mastering
engineers will normally do a master recording at a higher sampling rate (i.e.
88.2, 96, 176.4 or 192 kHz) and then do any editing or mixing at that same
higher frequency. High resolution PCM recordings have been released on
DVD-Audio (also known as DVD-A), DAD (Digital Audio Disc—which utilises the
stereo PCM audio tracks of a regular DVD), DualDisc (utilising the DVD-Audio
layer), or Blu-ray (Profile 3.0 is the Blu-ray audio standard, although as of
mid-2009 it is unclear whether this will ever really be used as an audio-only
format). In addition it is nowadays also possible and common to release a high
resolution recording directly as either an uncompressed WAV or lossless
compressed FLAC file[8] (usually at 24 bits) without down-converting it .
However if a CD (the CD Red Book standard is 44.1 kHz 16 bit) is
to be made from a recording, then doing the initial recording using a sampling
rate of 44.1 kHz is obviously one approach. Another approach that is usually
preferred is to use a higher sample rate and then downsample to the final format's
sample rate. This is usually done as part of the mastering process. One
advantage to the latter approach is that way a high resolution recording can be
released, as well as a CD and/or lossy compressed file such as mp3—all from the
same master recording.- Wikipedia
Now that the theory has been covered by Wikipedia, I’ll continue.
Personally, I couldn’t disagree more with the idea of mp3 files as
the accepted standard of audio distribution. This is because, despite the
inherent benefits of these highly compressed files, such as the ability to
store tens of songs in the place of just one lossless “CD-quality” audio file,
the algorithm used to compress them to this extent is flawed, and I feel
somewhat disingenuous.
This is because it uses what is known a “lossy” compression
method. This means that data is actually discarded from the original file in
order to achieve optimal size. It does this by using a technique called
perceptual coding, which discards or reduces the precision of sound components
less audible to human hearing, and then records the remaining information in a
fairly efficient manner. At the standard sample rates of 128, 256 and 320 kbps,
the result is a file which on the face of it sounds almost identical to the
original. It will almost certainly not be distinguishable through the average
stock iPod headphones. However, on closer inspection through a decent setup and
professional quality headphones or speakers, many files will begin to show
differences. The files can often sound very synthetic, and have their very own
signature sound. In fact, many actually prefer the distinctive punchy sound the
compression can produce, and more still simply cannot tell the two apart.
The degree to which the compression method harms the
original file can depend on the genre of the music, its complexity and the standard
to which it was produced to begin with. For example, modern electronic music
will often respond better than classical full orchestra piece.
Beyond this perceived audible difference of each file though,
is the matter of principle. After the artists have worked tirelessly in the
studio to perform every second of a song to perfection, the raw recording is
passed on to a very clever and talented professional who will spend days,
weeks, or months tweaking every frequency and channel to a level suitable to
complement each element of the piece to the best of their ability. As a result,
I feel that to then take this hard work and expertise and soil it by discarding
roughly an 1/11th of the data, altering the EQ with gimmicks like
“bass-boost”, and then proceed to listen to it through headphones with the
audio response of two potatoes connected with some rusty fence wire, is simply
wrong.
I’ll admit though, until roughly 4 years ago, I had never
really questioned “lossy” compression or audio quality in general. I would rip
all my CDs in the default compression method of a player and listen to it
through my mp3 players bundled plastic soul destroyers. The changing point was
when I tried my older brother’s Sennheiser HD 25 1 II headphones. These are pretty much regarded worldwide as the standard
for professional sound monitoring. Unlike many headphones, they produce a very
precise, flat response with controlled treble, defined mid and tight bass, thus
meaning that the recording will sound almost exactly as the artist intended.
Even connected to my old 1 GB Zen Stone, I was astounded by just how different
my music sounded. Of course, being the naïve teenager I was, the first thing I
noticed was the bass; something which most standard in-ear buds struggle to
produce faithfully. I listened through all my favourite songs, falling in love
quickly with the new, more dynamic sound; I was smitten. Eventually I got hold
of a pair for myself, and it wasn’t long before their clean and precise
response had me noticing the flaws in the mp3 files, and my entire library of
music was re-ripped in a lossless format.
I’ve been through many different
lossless file types: like WAV (the audio equivalent of a bitmap image file), FLAC
(a lossless compression which uses a clever algorithm to compress audio data to
around half its original size without loss of data), and now ALAC (Apple’s
version of FLAC). I currently own a 160 Gb iPod Classic which is around 2/3
full of music encoded in lossless ALAC, which drives my HD 25s. When I buy an
album now, I can be sure that I’m hearing it as it was intended to be heard, as
and I honestly cannot see why anyone who truly loves music would do otherwise.
Of course, this is just my opinion, and while many may agree with
me, many more will disagree with me. I’m not
trying to brainwash you or to tell you that you’re wrong to use mp3s, or
standard in-ear buds. I’m merely suggesting that you try the alternative, if you
can. And if you enjoy it, you’re welcome, if not, I’m sorry for wasting your
time.
- Michael Palmer
If there is anything negative to be said about this record, it’s that, while there have been no real steps backwards for the group, it doesn’t seem to build upon the band’s discography as much as their past releases have. In other words, L’Enfant Sauvage brings practically nothing new to the table for the band, musically. While this isn’t necessarily a bad thing for most fans of the group, if you’re like me, you may have grown weary of the Gojira’s sound from extensive listening to their past efforts. While that may reflect personal change more than it does the musical prowess of the band, this past album seems to exhibit less development than any of the group’s previous releases have, and, because of this lack of evolution, listening to L’Enfant Sauvage may not be anywhere near as mind-blowing as their past two releases were the first go-round if you are already familiar with the work of Gojira.
