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DIGITAL SOUND
In order to create the best possible sounds of your own, it is
important to know something about digital sound. Here we will try to
explain some things which will hopefully help you a lot.
example of
digital sound
Please Play digital sound demo
file after complete opening of this page
Floating
Clouds.mp3
Sitar.mp3
This document has been divided into several separate parts:
The theory of digital sound
Basic signal theory
As you probably know, sound is air which is moving very quickly. The
speed of these movements is called "frequency", which is a very
important property of sound, especially music. The frequency of a
sound is measured in Hz (=Hertz, named after a man called Hertz :-/
who did a lot of research into sound and acoustics some time ago).
Most people can hear frequencies in the range between 100Hz-15000Hz.
Some people can hear very high frequencies above 19000Hz, but
scientists always assume that the human ear is able to discern
frequencies between 20Hz-20000Hz, since those numbers make their
calculations a lot easier.
Here's a few examples of different frequencies, if you'd like to play
with them for a while:
60 Hz 440 Hz 4000Hz 12000Hz
15000Hz 20000Hz
-very- low medium high very high
Another very important property of sound is its level; most people
call it volume. It is measured in dB (=deciBell)
Most professional audio equipment uses a VU meter (=Volume Unit meter)
which shows you the input or output level of your equipment. This is
very convenient, but only if you know how to use it: A general rule is
to set up the input and output levels of your equipment so that the
loudest part of the piece you want to record/play approaches the 0dB
lights. It is important to stay on the lower side of 0dB, because if
you don't, your sound will be distorted badly and there's no way to
restore that. If you're recording to (analog!) tape, instead of
(digital) hard disk, you can increase the levels a bit, there is enough
so-called 'headroom' (=ability to amplify a little more without
distortion) to push the VU-meters to +6dB. There is some more
information on calibrating equipment levels in the recording section
below.
Some examples of different levels, if you'd like to play with them for
a while:
0,0dB = 100%
-6,0dB = 50,0%
-18,0dB = 12,5%
+6,0dB = 200%
maximum level
half power
very quiet
a little too loud-a lot of distortion
Okay, now that you know the most important things about sound, let's
finally go to the digital bit (ooh, a pun :-/ ): I've just told you
about the properties of 'normal' (analog) sound. Now I'll tell you
what the most important properties of digital sound are.
Digital Audio Theory
The sample rate of a piece of
digital audio is defined as 'the number of samples recorded per
second'. Sample rates are measured in Hz, or kHz (kilohertz, a
thousand samples per second). The most common sample rates used in
multimedia applications are:
8000 Hz
12000 Hz
22000 Hz
really yucky
not much better
only use it if you have to
Professionals use higher rates:
32000 Hz
44100 Hz
48000 Hz
only a couple of old samplers
Nice, what a relief
some audio cards, DAT recorders
Some modern equipment has the processing power required to enable even
higher rates: 96000Hz or even an awesome 192.000Hz will possibly /
probably be the professional (DVD?) standard rates in couple of years.
The advantages of a higher sample rate are simple: increased sound
quality. The disadvantages are also simple: a sample with a higher
sample rate requires an awful lot more disk space than a low-rate
sample. But with the hard disk and CD-R prices of today that isn't too
much of a problem anymore.
When recording a certain frequency, you will need at least (but
preferably more than) two samples for each period, to accurately
record it's peak and valley. This means you will need a sample rate
which is at least (more than) twice as high as the highest frequency
you'd like to record, which, for humans, is around 20000Hz. That's why
the pro's use 44100Hz or higher as the minimum sample rate! They can
record frequencies up to 22050Hz with that. (Now you know why an 8000
Hz sample sounds so horrible: it only plays back a tiny part of what
we can hear!)
Using an even higher sample rate, like 96000Hz, you can record higher
frequencies, but you won't hear things like 48000Hz anyway. That's not
the main goal of those super-rates. If you record at 96000Hz, you will
have more than four samples for each 20000Hz period, so the chance of
losing high frequencies will decrease dramatically! It will take quite
a few years for consumer level soundcards to support these numbers,
though. There are a few pro cards which already do, but you could
easily buy a small car for the same money...
That's enough about frequency for now. As I said before, another very
important property of sound is its level. Let's have a look at how
digital audio cards process the sound levels.
Dynamic range
The capacity of digital audio cards is measured in bits, e.g. 8-bit
soundcards, 16-bit soundcards. The number of bits a sound cards can
manage tells you something about how accurately it can record sound:
it tells you how many differences it can detect. Each extra bit on a
sound cards gives you another 6dB of accurately represented sound
(Why? Well, Because. It's just a way of nature). This means 8-bit
soundcards have a dynamic range(=difference between the softest
possible signal and the loudest possible signal) of 8x6dB=48dB. Not a
lot, since people can hear up to 120dB. So, people invented 16-bit
audio, which gives us 16x6dB=96dB. That's still not 120dB, but as you
know, CD's sound really good, compared to tapes. Some freaks, that's
including myself ;-) want to be able to make full use of the ear's
potentials by spending money on soundcards with 18-bit, 20-bit, or
even 24-bit or 32-bit ADC's (Analog to Digital Converters, the gadgets
that create the actual sample) which gives them dynamic ranges of
108dB, 120dB, or even 144dB or 192dB.
Unfortunately, all of the dynamic ranges I mentioned are strictly
theoretical maximum levels. There's absolutely not a way in the world
you'll get 96dB out of a standard 16-bit multimedia sound card!!! Most
professional audio card manufacturers are quite proud of a dynamic
range over 90 dB on a 16bit audio card. This is partly because of the
fact that it's not that easy to put a lot of electronic components on
a small area without a lot of different physical laws trying to get
attention. Induction, conduction or even bad connections or (very
likely) cheap components simply aren't very friendly to the dynamic
range and overall quality of a soundcard. But there's another problem,
that will become clear in the next paragraph.
Quantization noise
Back in the old days, when the first digital piano's were put on the
market, (most of us didn't even live yet) nobody really wanted them.
Why not? Such a cool and modern instrument, and you coould even choose
a different piano sound!
The problem with those things was that they weren't as sophisticated
as today's digital music equipment. Mainly because they didn't feature
as many bits (and so they weren't even half as dynamic as the real
thing) but also because they had a very clearly rough edge at the end
of the samples.
quantization noise Imagine a piano sample like the one you see here.
It slowly fades out until you here nothing.
At least, that's what you'll want... As you can see by looking at the
two separate images, that's not at all what you get... These images
both are extreme close-ups of the same area of the original piano
sample. The highest image could be the soft end of a piano tone. The
lowest image however looks more like morse code than a piano sample!
the sample has been converted to 8 bit, which leaves only 256 levels
instead of the original 65536. The result is devastating.
Imagine playing the digital piano in a very soft and subtle way,
what'd you get? some futuristic composition for square waves! That's
not what you paid for ;-) This froth is called quantization noise,
because it is noise that is generated by (bad) quantization.
There is a way to prevent this from happening, though. While sampling
the piano, the soundcard can add a little noise to the signal (about
3-6dB, that's literally a bit of noise) which will help the signal to
become a little louder. That way, it might just be big enough to get a
little more realistic variation instead of a square wave. The funny
part is that you won't hear the noise, because it's so soft and it
doesn't change as much as the recorded signal, so your ears
automatically forget it. This technique is called dithering. It is
also used in some graphics programs e.g. for resizing an image.
Jitter
Another problem with digital audio equipment, is called jitter. Until
now, I've always assumed that the soundcard recorded the sample at
exactly 44100Hz, taking one sample every 1/44100 second. Unfortunately
that is -totally- unreal. There *always* is a tiny timing error which
causes the sample to be taken just a little too late or just a little
too soon.
Does this make a big difference then? Well, you could start nagging
about everything, but then you'd probably have bought a more expensive
soundcard in the first place. The really bad part is that jitter is
frequency dependent. Because it's related to the timing of the sample,
it can change the recorded frequencies just a little. If it records a
sample just a little too soon, the card thinks that the recorded
frequency is a little lower than it really is. This is noticeable at
frequencies below 5000Hz but especially bad at the lowest frequencies,
because the influence of a little error is much bigger there. Typical
jitter-times go between 1.0 x 10 -9 seconds (that's a NANO second,
read: almost nothing) and 1.0 x 10 -7 seconds (that's a hundred
NANO seconds, not a lot more) but they make the difference between a
'pro' sound and a 'consumer' sound on e.g. different CD-players.
Digitizing sound
When you record a sample with your sound card, it goes through a lot
of stages before you can store it on your hard disk as a sound file.
Fortunately you don't have to worry about these stages, because modern
sound cards and samplers take care of them for you.
I'm going to be a big bore and tell you about these stages anyway.
Let's see what happens when you press 'rec':
The sound card starts a very accurate stopwatch (the sample rate).
AD conversion process
Analog to Digital Conversion process
Then it transforms the sound coming in: it simply cuts off the very
high frequencies which it cannot handle. This cripples the sound a
lot, but it is required to prevent even more serious damage to the
sound, which would make the sound unrecognizable. This is a low-pass
(cut the 'high' frequencies, let the 'low' frequencies pass through)
anti-aliasing (smoothing, blurring) filter (because it takes away some
parts and leaves the rest)
Every time the stopwatch has completed a cycle, the sound card's ADC
looks at the filtered input signal. It calculates how loud the
incoming sound is at that exact moment in time (very much like a
microphone would measure air pressure) and transforms the loudness
level into the nearest digital number.
and shouts that number to the computer, which stores it somewhere in
memory, probably on a hard disk.
Sound card manufacturers put a brickwall-filter (look at the image
below!) in their sound card, to prevent a very nasty side-effect
called 'foldover'. Foldover is a pretty difficult concept, but I'll
try to keep it simple.
It's more or less the same thing that happens when you look at a car's
wheel when it drives past you very quickly. You'll sometimes see the
wheel moving backwards. Another example can be found in old western
movies where you'll see a train going by. The 'wheels' of the train
will be moving backwards too, if the train's going fast enough.
All these 'illusions' are foldover-effects. They occur when a fast
system at regular intervals analyzes something which is moving even
faster than the system itself.
When recording at 22050Hz, your sound card will simply not be able to
record any frequencies above 11025Hz, because you need at least two
samples for each period, as described above. Without the low-pass
filter, the sound card would blindly try to record those frequencies.
But afterwards, when you play back the sample, you'll hear a totally
different frequency instead of the original one. Just like the car's
wheel that seems to be moving backwards, while it really isn't.
(The frequency you'll actually hear equals the sampling frequency
minus the original frequency, e.g. 22050-12050=10000Hz, instead of the
original frequency, in this case 12050Hz).
'brickwall' filter
a brickwall filter at 4000Hz
Therefore, the maximum frequency that can be recorded with a certain
sample rate, is half the sample rate. That frequency is called the
Nyquist frequency, sometimes abbreviated to fN, after a man named
Harold Nyquist, who worked at Bell Telephone Laboratories and more or
less invented audio sampling. A big guy in digital audio. Anyway, to
prevent all that from happening, the sound card manufacturers put a
special filter in their card (see figure of brickwall filter on the
right).
This low-pass filter removes high frequencies like any equalizer or
Hi-Cut Switch does, except it is *much* more agressive. You can see
that the filter allows all sound below 1000Hz to pass through, and
that it gives the frequency range of 1000Hz-3500Hz a small boost.
(This boost is necessary to be able to cut off the higher frequencies
with such violence.) Frequencies above 4000Hz are eliminated extremely
agressively. That is why they call it a brickwall-filter, because of
the wall-like slope.
The filter displayed above might be used for a sample rate of about
8000Hz, since an 8000Hz sample has a Nyquist frequency, the maximum
recordable frequency, of 4000Hz. This makes it very important to
choose the appropriate sample rate for your sample; that is, if you've
got a legitimate reason not to record at 44100Hz, or higher ;-)
Recording digital sound of your own
Let's go through this step by step.
We'll start by selecting File->New, something which every sample editor
I know can handle ;-). You'll want to select the number of bits you'll
want to use for each sample. You'll also want to select the sample
rate. My advice is: pick the highest your hardware can handle. That is
most likely 16 bits at 44100Hz, since most, if not all, consumer sound
cards support CD-quality playback & record.
VU-meter VU-too low VU-Clip
Then let the band, or whatever, play for a while, to see if you're
recording levels aren't too high or too low. Your program probably
supports input monitoring and If if yours doesn't, it should! Get
yourself another program ;-) You'll probably see a variant of the good
ole VU-meter I like the one to the right. The loudest part of the
sound you want to capture to disk should be somewhere very near 0.0dB,
but it should not, ever, never ever!! exceed 0.0dB, since that results
in very nasty distortion, which is cool on analog recorders but really
horrible in the digital world.
If you want that distortion effect, get a program to do it for you,
but don't record at a too high level! Sonic Foundry's Sound Forge has
a really good Distortion feature. Also, there are lots of Direct-X
plug-ins which emulate tube compression and tape saturation etc. This
type of digital distortion is called 'clipping' because all samples
that exceed the maximum level are 'clipped', (cut off and reduced) to
the maximum level.
Don't set your recording levels too low, though. It will further reduce
the accuracy of your home recording, since multimedia cards already add
a very significant bit of noise. In fact, they sometimes hardly leave
you any dynamic range at all!
So, be very picky about your input levels.
Next, think about the source of your recording. A microphone? A
keyboard or synthesizer? a DAT-tape? If the source already is digital,
like with DAT and CD, please go ahead and stay digital! Use a digital
connection between the DAT and the soundcard, to prevent the operation
of digital-to-analog conversion -> transmission through a cheap cable
-> analog-to-digital conversion from adding noise or distortion!
If you're recording with a microphone, first let the microphone record
a minute or so of 'silence'. Then play that recorded 'silence' back
over headphones and listen the amount of noise coming from the room.
Be sure to keep this data, because some good programs can eliminate
that noise from the actual recording, by using the data as a 'noise
print' (They analyze the noise print data and then 'subtract' it from
the real recording. Sound Forge and CoolEdit have this great feature.)
Also, if you have the opportunity, try several different microphones
for the same recording. Learn to trust your ears. If you have several
different recordings of the same event, pick the one that sounds best.
Don't automatically pick the one recorded by the most expensive mic.
That! Does! Not! Work! Pick the one that sounds best. You'll be
surprised to hear the number of top hits being recorded with cheap
mics. But I'm not saying you should be using cheap mics... There are
several pretty good all-round microphones available from $30 (like the
Behringer XM-2000). A really good mic for vocals and guitar is the
SM-58 by Shure. These are a little more expensive (over $100), but
they are used all over the world in pro studio's. The problem with
these microphones is, you'll need a pre-amplifier too, because the
original microphone signal is very weak, and an 'XLR-cable' to connect
it to your gear. Most mixers have microphone pre-amps on them. If
you're looking for a good value-for-money mixer, I suggest you take a
look at Behringer's website. They're not 100% top quality, but if 90%
is good enough for you (It's that last 10% of perfection which makes
audio equipment so darn expensive) Behringer is the place to be.
If you're recording from a different piece of hardware e.g. directly
from synthesizer/keyboard, check your manual to see if your hardware
has balanced outputs. If it does, you'll need to get/make two stereo
jack plugs and three wires of the same length, or even better: an
insulated cable with three separately insulated wires (that's a
multi-buck issue, though...) to make sure your audio isn't distorted
before it goes into your sound card's inputs.
A normal wire has 1) a signal wire and 2) a ground wire. If you use
normal wire over long distances, preferably close to stage lighting
;-) you'll notice the wire picks up an awful lot of noise and buzzing
on the way. This has something to do with induction and magnetic
fields but all you'll need to know is that it sucks. To prevent such
50Hz (AC power!) buzzing, the professionals use balanced cables.
The balanced cable system is a very nice way of connecting equipment
over long distances without loss of sound quality or unwanted
induction. This is possible because a balanced cable has three wires
instead of two: 1) a signal wire, 2) an inverted signal wire and 3) a
ground wire. At the output of the synthesizer / mixer / whatever, the
output signal is routed to both the signal-wire and the
inverted-signal-wire.
The signal going to the inverted signal wire is then inverted
(multiplied by -1, turned upside down, given a phase-shift of 180
degrees) and transported together with the signal wire all the way
through the cable to the other connector and on the way, both wires
pick up all the usual noise and hums. But when the signal arrives at
its destination, the inverted signal is inverted again, so that the
signal it was carrying is back to normal again. But this inversion
also inverts the noise and buzz, so now we have: a signal wire with 1)
the signal and 2) the noise, and we have the re-inverted(=normal!)
wire with 1)the signal and 2)the inverted noise. These two are mixed
together by the equipment: signal + signal + noise - noise, which
gives twice the signal strength and no noise whatsoever!
i
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Basics
Audio Formats
Your computer will save audio information to its hard drive (or
specified floppy disk drive) in a digital format depending on what
type of hardware, operating system and audio-related application you
are using. These digital audio sound files do not degrade with
continuous use and listening. You can listen to these files on your
PC, either from a file, CD or over the Internet, depending on what
format they are in and what type of audio-related application or
player your computer has installed on it. These formats are identified
by their file extension. There are many different sound files type
that have developed over the years due to various competing hardware,
operating systems and software applications. These sound files differ
from each other based on the encoding parametrs (how the actual sounds
were stored in the file) and their compression formats. Some files
even include additional information other than the digital audio data.
Some of these formats have been superseded by newer formats and are
not widely in use any more but you will still see references made to
them.
Format, File Extension or Acronym Association
AAC MPEG-2 Advanced Audio Coding video and sound file
ACD Acidized Sonic Foundry Project (.wav files) File (.acd)
ADM Adaptive Delta Modulation
ADPCM Adaptive Differential Pulse Code Modulation
AIFF Audio Interchange / Information File Format (Apple Computers, .aiff,
.aif and .aifc compressed format)
APE Monkey's Audio digital audio file (.ape)
ARR Steinberg Cubasis AV Arrangement File
ASF Microsoft Active Streaming Format (.asf)
AVI Microsoft Audio Video Interleave File
AVR Audio Visual Reasearch, Atari computers
BWF Broadcast Wave format subset specified by the European Broadcast
Union
CBK Creative Labs Wave Blaster Sound Bank file
CDA Compact Disk Audio (sometimes as CD-DA)
CFX Crystal Audio Engine Audio real-time effects plug-in
CYC Cakewalk Cyclone Groove Sampler embedded sample and settings File
CWO Crystal Audio Engine Audio Output
DLS Down Loadable Sounds (.dls)
DS DrumSynth .ds file
DSP Dynamic Studio Professional Module
DTS Digital Theater Systems
FLA Macromedia Flash File
FLAC FLAC (Free Lossless Audio Codec) lossless digital audio file
(compatible with Ogg Vorbis)
F3r Farandole Blocked Linear Module
FXP Steinberg HALion instrument file (.fxp)
GIG GigaStudio Native Instrument file (.gig)
GKH Ensoniq Disk Image (VFX, SD, EPS, ASR, TS)
GS MIDI Roland Instruments MIDI
IBK Instrument Bank File (FM sysnthesis)
IDF MS Windows Instrument Definition File
IFF Interchange Format File (IFF), Amiga Computers format
IMA ADPCM Interactive Multimedia Association Adaptive Differential
Pulse Code Modulation
INS Cakewalk Instrument Definition File
INS Ensoniq Instrument File
KMP Korg Trinity KeyMap File
KRZ Kurzweil K2000 File
MOV Apple Quicktime and QDesign multimedia File (.mov)
MPE Destiny MPE Secure Audio
MPEG-1 MPEG (Moving Pictures Experts Group) Layer 1 (.mp1)
MPEG-2 MPEG (Moving Pictures Experts Group) Layer 2 (.mp2)
MP3 MPEG (Moving Pictures Experts Group) Layer 3 (.mp3)
MPEG-4 MPEG (Moving Pictures Experts Group) 4 File (.mp4, .mpe)
MPG Moving Pictures Experts Group
MPGA MPEG Audio File (.mpga)
MTM MultiTracker Module
MT2 Mad Tracker 2 Sound Module
M3U .mp3 Playlist File (.m3u)
MX3 MPEG File (.mx3)
OGG Ogg Vorbis File (.ogg)
PCA Perfect Clarity Audio (Sonic Foundry)
PCM Pulse Code Modulation (Uncompressed Raw Sound File) (.pcm)
RA RealAudio (.ra)
RAM RealAudio Meta-file for redirecting to an RM streaming format file
on an HTTP or Real server (.ram, also .rmm)
RAW PCM Signed Raw Sound File
RIFF Resource Interchange Format File, WAV is a subset of this format
RJM RealAudio 8 File (.rjm, Native format from RealNetworks)
RM RealMedia File (.rm; also .rmx, .rmj, .rms)
RMI An RIFF MIDI format that includes copyright and non performance
data (MS Windows)
ple file
SACD Sony/Philips Super Audio Compact Disk
SBI Creative Labs Sound Blaster Instrument file
SDII Sound Designer II by Digidesign (also as .sd2 or Session8 on
Windows)
SF2 SoundFont, version 2 (EMU-Systems) (.sf2)
SFD SoundStage Sound File Data
SFR Sonic Foundry
SOU SBStudio II Sound File
S22 Digigram; also S32
SYX Yamaha XG compatible project file (saving and loading)
SYX MIDI System Exclusive File
VBA VBase ADPCM File
VCE Natural MicroSystems VCE File
VOC Creative Lab's SoundBlaster/Creative Voice Sound File (.voc)
VOX Dialogic Vox ADPCM File
VOX Pika ADPCM File
VQF Yamaha Sound TwinVQ (.vqf)
WAV Microsoft Windows file format, RIFF Waveform Audio (.wav)
WMA Microsoft Windows Media Audio (.wma)
XG MIDI Yamaha instruments MIDI
XGP Yamaha XG parameter files (for Loading only)
Structured Audio
Structured Audio is synthesized sound that is created by program
applications that can process a alphanumeric/text language that
describes the sound, and then direct the synthesizer to produce an
audible sound. The languages allow for the construction of very
complicated instruments.
CSound
CSound is a music programming / descriptive language. It is supported
by Windows and Unix-based operating systems. Sound is produced through
construction of sound sysnthesis components by CSound language code,
which is entered into a text editor. The composer can write the
performance and describe the qualities of the instruments in the
performance. The instrument description is saved in a file and then
the instrument can be played by MIDI controller. The application is
presently in the public domain and is constantly being modified.
SAOL (Structured Audio Orchestra Language) and SASL (Structured Audio
Score Language)
SAOL is a language that can be used to describe musical instrument
parameters again based on creating sound synthesis components by
mathematical equation. SASL is the companion language that controls
the instruments created in SAOL. SAOL is incorporated into the MPEG-4
codec and is flexible enough to describe sysnthetic sound generation
(additive, physical modeling, etc.) or manipulate a bank of digital
audio sample files. SASL is the actual descriptive language that
defines instruments and/or the entire performance by incorporating its
data in the bit stream of the codec.
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How to Properly Encode Dolby Digital Audio (AC3)
Introduction
Many people on the forum have experienced problems when encoding audio
using Dolby Digital. These problems are primarily volume-related, with
some dynamic range compression issues as well. This guide aims to
educate about how Dolby Digital audio should be encoded, and how to
make it sound best.
References
The primary references for the information contained in this guide are
two guides on Dolby's web site. The first is Standards and Practices
for Authoring Dolby Digital and Dolby E Bitstreams, which has the best
information on Dynamic Range Compression. The other is Dolby Digital
Professional Encoding Guidelines which gives an excellent explanation
of the dialogue Normalization parameter. You will need Adobe Acrobat
Reader to view these .pdf documents.
Philosophy of Dolby Digital
Dolby Labs has been doing high-quality audio with cutting-edge
techniques for a long time, using their past experience as a guide. As
such, there is often confusion about their methods and philosophy to
those of us who are not privy to that information. Of prime example is
the current problem: Why is Dolby Digital so much quieter compared to
my original sound?
Most audio destined for DVDs is audio originally recorded for use in
the movie theater. The movie industry has a huge advantage when
producing audio for the theater -- the theater has large speakers and
amplifiers, and a quiet, near-ideal listening environment. Huge
dynamic ranges are possible, where the slightest whisper of dialogue
is audible, yet gunshots and explosions can be earth-shattering.
Dolby's dilemma was: "How do we bring this audio, with its huge
dynamic range, into the home?" This is a major problem -- most homes
don't have the speakers and amplifiers necessary to shake the living
room. Further, background noise in the home can easily drown out those
subtleties in the soundtrack.
Dolby's answer is to allow the decoder to modify the sound to
compensate for these problems. Low-volume sounds are boosted
automatically so they can be heard, whereas high-volume sounds are
quieted down so that speakers aren't blown and other persons in the
home are not disturbed. Further, Dolby Digital allows for different
program material to be equalized, so that volume does not have to be
adjusted when switching between inherently quiet programs and
inherently loud programs.
Decoder Specifics
The methods I'm about to present here for encoding Dolby Digital are
generic and do not apply specifically to any one encoder. All
Dolby-certified encoders (and some non-certified ones) will have the
appropriate parameters available to follow this procedure. I have
personally tested the Sonic Foundry 5.1 Plug-In Pack for ACID Pro, as
well as Sonic Foundry Soft Encode. These methods should also work for
BeSweet, Vegas Video + DVD, Scenarist, and other software-based
encoders.
Basic Parameters
Every Dolby Digital encoder has some basic parameters that need to be
set. |
