her they might have cows seemed from
Andrew authority now audio is a big part
of this smartphone experience and people
spend time and money investing in
headphones and wireless headphones and
speakers to try to improve that audio
experience but of course if the source
of your audio is bad then doesn't matter
how expensive your equipment is you're
still going to get bad quality sound so
the question before today is how do
computers store audio well let me
explain in the real world sound waves
travel through the air they arrive at
our ears and our brain interprets them
into music and speech and other types of
sound now we can see those sound waves
when we register them on a computer now
the biggest difference between the
analog world and the digital world is a
digital world works on timeslots
everything happens in a certain time
frame now those time slots might be very
very tiny however they are still time
thoughts they are not continuous so when
we record sound we need to ask ourselves
the question how often should we be
recording the level of the sound wave
the amplitude of the sound wave let me
give an example if I said to you I
wonder if you could tell me in an email
the temperature throughout the day
outside your house now you could just go
outside your house once look at the
temperature and come back and send me a
number 23 degrees Celsius well that
doesn't tell me how much it is through
the day I've lost a lot of information
there so I say no can I have something
that's a bit more accurate please so
maybe you go out and you register it in
the morning at lunchtime and in the
evening but again that's only three
points of data and it's a very very
rough so we could say where they measure
it every hour now that would give me
maybe a better graph could you measure
the temperature every 10 minutes could
you measure it every minute could you
measure it every second so you see as we
increase the accuracy we do it more and
more often
that's exactly the same with sound you
could measure it once over a whole
second and you would get no information
at all you'll just be be rubbish you
could start to measure it quicker and
quicker and quicker and then you get a
better representation of the sound wave
now in audio this is
the sampling rate how often do you take
a sample to see the level of the
Soundwave now there are some
mathematical theorems that come in here
and those important is one called the
Nyquist theorem which tells us that to
register something we need to use twice
the frequency of the sound wave that
we're measuring now the human ear can
hear up to about 20 kilohertz so we need
to use 40 kilohertz as our sampling rate
as a bare minimum
now cd-quality was 44 point 1 kilohertz
and today some systems use 48 kilohertz
and those are basically the standard
sampling rates for music now there are
some situations where people record in a
higher sampling rate now I'm talking
here about playback through your
smartphone or maybe through some other
audio equipment if you're doing studio
work there may be arguments for
recording for sampling at a higher rate
but for playback 44.1 or 48 kilohertz
are just absolutely fine now the other
side of this is you say well what great
what measure what gradient are we using
to measure this sound wave again I could
say to you please go outside every hour
and tell me the temperature and you
could come back and say it was hot and
then it was hot and then it was cold
well that doesn't really tell me very
much that's just binary information hot
or code one bit of information so maybe
you could increase your scale maybe in
the morning you could come out and say
it was very cold by mid morning it was
just cold during the afternoon it was
hot then late afternoon it was very hot
and then he got cold again and then very
cold and here I just have four different
states that gives me two bits of
information well of course if you think
about a thermometer when we're measuring
temperature maybe there's a hundred or
120 130 different levels that we can
measure temperature on maybe even more
if we use fractions so with audio we
need to have a good system for measuring
the level of the audio now eight bits
would give us 256 different levels and
sixteen bits
give us over 65,000 different levels and
24-bit gives us over 16 million
different levels now there are some
arguments for using 24-bit audio and
we'll go into that in a minute
however a lot of systems use just 16-bit
audio which gives you 65,000 different
levels when you are reaching the sound
so at forty four thousand times a second
a point is plotted on a graph somewhere
within a range of 65,000 different
points and that is how the sound is
recorded and that system is called PCM
pulse code modulation now when you have
this digital music on your smartphone it
needs to be turned back from a digital
system into an analogue system but then
powers your headphones or a speaker or
something else now to do that you need
to use something that's called a digital
to audio converter and the digital to
audio converter the DAC has the job of
taking all those reference points of
information about the waves in the sound
and converting it back again into real
sound
now there are lots of different
technologies involved in doing that
however there are a covering important
things to realize one is that sometimes
you might see diagrams that show kind of
the waves as being squares with big
steps on them well that's actually not
quite true the way DAX work is using
interpolation using some filters they
are actually able to smooth out the data
the sine wave that comes out from that
data one of the ways they do that is
using over sampling them every time you
over sample something you can actually
if you double the over sample you can
reduce the amount of noise in the
audible spectrum by up to 3 dB so
actually some - we'll go do lot over
sampling and then reduce it back down
again to produce the sound wave so
there's lots of technologies involved
but don't think that Dax are producing
kind of these square waves they're not
it's all very very smooth now your
mobile phone will have a DAC in it and
hopefully it will have a high-quality
DAC and hopefully it will produce good
sound from that digital audio of course
the cheaper the phone the charts are the
cheaper the component and the chance to
have a cheaper the DAC that's why you
need to be happy with the DAC that's
inside your phone now I mentioned
earlier on there was an argument for
using 24-bit playback and 24-bit
recording now the reason behind it is
this all audio circuits produce an
amount of
now the amount of noise they produce
depend on the quality and so on now the
best we can produce a day is a hundred
and twenty four decibels of
signal-to-noise ratio now one hundred
and twenty four decibels means 21 bits
of information now 21 bits is greater
than the 16 bits you find a lot of
format and it's coming close to a
hundred and two twenty four bits so
24-bit would seem to be the optimum best
situation for audio playback now I'm not
talking about studio stuff here if
you're doing things in the studio there
is a good argument for using 32 bits
because sound waves need to be
manipulated they need to be added they
need to be changed need to be mixed
around and you need a lot of bandwidth
so there's no clipping going on I'm
talking now about playback 24 bit
playback really is the best that we can
expect now the problem with a PCM format
this RAW format of always data with
capture those are registering the sound
wave is it can produce very large files
for example a 16-bit capture of four
minutes of music at 16 bits 44.1
kilohertz will produce a file size of
around 40 megabytes now key that's not
good for streaming services it's not
good when you're streaming data over 3G
or 4G so there has to be a way of
generating smaller files and that's
where we get into the different file
formats that are available today now
there are two different types of file
format one is called lossless which
means there is no loss of any of that
data that was used to record the sound
originally and the other is lossy which
means that there has been some quality
lost during the production of the sound
file now the wave dot wav files you
might find on PCs it's really a raw PCM
format and that is lost less there's
nothing lost in that there's also this
very popular codec called flak and that
also is lossless now flak has advantage
that why it doesn't lose any of the data
in the file it does use compression
which means it can shrink down file size
to roughly a half however that's still
pretty big so then we move on to the
lossy formats now the way these work for
example MPs
is a classic example of a lossy format
is there are algorithms that are used
trying to understand how the brain works
trying to understand how the ear
interprets these sounds and chops out
bits that they reckon can't be heard now
of course those that are very sensitive
about music will say but you can hear it
there is a big difference and I'm not
going to get into that argument but the
idea of a loss lossy compression
algorithm is it doesn't just strip away
randomly things it tries to strip away
things that are not needed and then it
also uses compression on top of that to
reduce the file size so for example four
minutes of wife might be 40 megabytes
but four minutes of an mp3 at 320
kilobits per second might only be nine
megabytes of ten megabytes so that's
really like a quarter of the size and
that's why mp3 is soap-opera today
because we can have relatively high
quality music near CD quality music that
is actually a lot smaller which is great
for streaming and great for storing on
our devices now there are other lost
lossy formats other than just mp3
there's also org forbus which is an open
source codec and there's also the
advanced audio codec AAC and AAC Plus
which is used more predominantly by
Apple and within iTunes and so on
however Android can play AAC files that
don't have any DRM now the advantage of
AAC is at a lower bitrate it certainly
has a greater audio quality than mp3
files a higher bit rates there are
arguments between people about which
one's actually best probably AAC comes
out on top however at higher bit rates
they certainly are comparable with each
other so let's just sum up audio is
recorded by measuring the amplitude of a
wave at a certain time interval and we
measure that roughly nowadays forty-four
thousand times a second or forty-eight
thousand times a second the measure the
gradient is used to measure that is
16-bit is 65,000 different level 24-bit
sixteen million different levels and
that produces the accuracy of which
we're registering something that come
from the analog world into a digital
represent
now when that gets into your phone and
you want to turn it back into analog
again it goes through a DAC and the DAC
converts that data back into a sound
wave and it's got lots of clever
technology inside of it that does things
like smoothing and shaping that actually
makes the sound that comes out as close
as possible to the original and there
are different quality Dax and different
quality phones have different Dax in
them and they are able to produce
different qualities of sound and that's
basically based on the cost of the DAC
inside of that phone and finally there
are different formats now we have an old
saying in software engineering that if
you put rubbish in you're gonna get
rubbish out so if you start with a
rubbish audio file doesn't matter how
good your audio equipment is you can get
rubbish out the other end but if you
start with a good source then there's a
chance you're gonna get good output now
there are lossless and lossy codecs
lossless include flak and wave wave
files lossy include mp3 ogg vorbis and a
a C and a AC plus each have their own
characteristics but the amount of
compression they can do the amount of
data that's stripped out of them and the
final file size well my name is Gary sim
from Andrew authority I hope you enjoyed
this introduction to audio and as I've
said on other of my videos this only is
an introduction it can't be more than
that in just a few minutes or video but
I do hope you liked it if you did please
do give it a thumbs up also don't forget
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