I've decided a few things over the last few days.
- 1 To learn how to create, mix, record and produce with no instrument ability, from the PC
- 2 To write a blog on my progress that may help people like me
I think the first hurdle is trying to learn the techno lingo and what it means as in DAWs and VST's and VSTi's etc. I say this is the first hurdle because when you begin to look at the things you will need to create, mix, record and produce something purely from a pc and screen, you hit the lingo straight away on websites like Sony and Cubase etc. A bit like when you begin texting or chatting and using smileys. It means nothing to you.
So starting with the basics as I must, I could spend hours looking up the terms and putting them into standard lingo. But for the large part, I don't need to because some kind people have already done a lot of the ground work for me such as Desktop_Music Handbook Dictionary.
The net is amazing when it comes to information but in this area, it's a little harder to get just what you want - a technical term dictionary for digital music production. Because production is not the sum of all the parts when it comes to technical terms and so we can have a list of terms for production but then the terms for mixing can be different (for example).
So the link above does not include the term DAW for starters. I wanted to attach a glossary I'd begun to build but can't so it's an incomplete but quite good list that adds to the above link.
Some Basic Audio and Recording Terms
analog audio. This is sound that is continuous, corresponding exactly to real sound.
This is unlike digital sound, which is broken down into a series of ones and zeros.
An analog signal is usually recorded using magnetic media, such as tape. Cassette
tapes and vinyl LPs, for example, both use analog sound. Analog recordings cannot
be copied without some loss of sound quality and are likely to contain more unwanted
noise than digital recordings.
ASIO. Audio Stream Input-Output. This is an audio protocol originally developed by the
Steinberg company and used by REAPER to communicate with professional-standard audio
hardware devices and interfaces (such as PCI soundcards and FireWire and USB devices).
bit depth (sample size). The bit depth is the level of detail at which a computer samples
analog audio to create digital audio. When recording WAV files, most commonly, you
use 16- or 24-bit sampling. 24-bit audio is generally preferred because it gives a more
accurate representation of the sound and makes it easier to avoid clipping, but it does
take up more disk space. Bit depth should not be confused with sampling rate.
bit rate. This measures the number of kilobits per second of data in MP3 and other
audio files. The bit rate you choose when creating an MP3 file determines the size
and quality of the resulting MP3. The highest commonly used bit rate is 320 kbps. A
file created using this bit rate will have excellent quality but will be fairly large. A standard
bit rate for encoding MP3s is 128 kbps. A file created using this bit rate will have
good quality and will take up about 1 MB per 1 minute of sound.
buffer. An audio buffer is a driver setting that helps determine the rate at which audio
passes between your computer’s processor and its soundcard. Reducing the buffer size
can help reduce the amount of latency while recording and monitoring audio. Increasing
the buffer size can help prevent pops and clicks while recording.
burn. This is the process of writing data or files onto a recordable CD using a hardware
device called a CD burner. Generally, you create either an audio or a data disc when you
burn a CD (although hybrid formats such as CD Extra are also available). Audio discs
can be played in any standard audio CD player. A data disc contains computer files and
can only be read on computers. Some DAW’s support the burning of audio discs.
bus. An internal pathway that may form part of your audio routing system. Although
REAPER makes no inherent distinction between a track and a bus, a track can be considered
to be functioning as a bus when more than one other track has been routed to it.
channel. A channel is a path through which an audio signal flows. One important feature
of REAPER is its ability to use up to 64 separate channels with any single track or
chorus. An effect that makes one voice or instrument sound like many. Chorus works
with all types of audio but is particularly effective with the human voice.
clipping. This is the unpleasant distortion you hear when output is too loud, causing
the peaks of the audio signal to rise above the capabilities of the amplifier circuit. To
avoid distortion, reduce the volume or gain before the stage in which the clipping
CODEC (or codec). Codec stands for compression/decompression. A codec is a program
used to enable Windows to compress and/or decompress audio to and from different
formats. For example, REAPER uses the LAME codec for creating MP3 files.
compression (audio). A process of reducing the dynamic range of an audio stream.
Often used to make the loud parts of a track quieter and the quiet parts louder. Not
to be confused with data compression, which is an entirely separate and unrelated topic.
compression (data). The process of packing digital data, such as computer files, more
efficiently for the purpose of storage or transmission. MP3 is one example of a commonly
used compressed audio format. Not to be confused with audio compression,
which is an entirely separate and unrelated topic.
control surface. An external device connected to your computer by MIDI cable, Fire-
Wire, or USB and that is used to physically control various parameters and functions of
decibel. A unit used for the measurement of sound. Commonly, sound pressure levels
(SPL) are represented as numbers from 0 dB (the softest sound that may be heard) to
120 dB and beyond (the level at which sound is perceived as pain). delay. An effect that
creates a delayed sound.
digital audio. Digital audio is audio that has been converted into a series of ones and
zeros that can be processed by a computer. When analog sound is converted in this way,
it is commonly saved and stored as a WAV file. Digital sound is easier to reproduce and
manipulate without loss in quality than analog.
digital audio workstation (DAW). A term used to describe a computer when it has been
set up and equipped with the necessary software and hardware to function as a recording
DirectX. A widely used plug-in format, generally less popular than VST.
Docker. A DAW’s Docker provides a tabbed viewing area for several DAW
functions (mixer, FX browser, undo, routing matrix, and so on) and is accessible
through the View menu. The Docker can often be detached from the main DAW window
and moved, for example, to a secondary monitor.
driver. Software that works with your computer’s operating system to control and use a
particular piece of hardware, such as your soundcard. To enable it to function at its
best, you should check the web regularly for any updated drivers for your soundcard.
dry. A term used to describe an audio signal to which no effects have been added. The
opposite is wet.
encoding/decoding. The process of converting audio to or from a compressed format,
such as MP3 or FLAC. The encoding and decoding processes require the use of codecs.
envelope. A technique used to control how any of a number of a track’s attributes
behave over time. For example, an envelope can be used to automatically fade a track’s
volume up and down as required for different parts of a song whenever it is played back.
equalization (EQ). Sound is made up of many vibrations that take place at the same time
at different frequencies. An audio equalizer lets you separately adjust the volume of
different ranges of frequencies, thus changing the makeup of the overall sound. Frequency
is measured in hertz (Hz). The higher the number, the higher the pitch of the
fade. A technique to bring sound into or out of a track gradually. Fade-in brings the
sound in gradually, fadeout does the opposite. Cross-fade occurs when two media items
overlap in such a way that one is faded in while the other is faded out.
fader. A device that enables you to control the level of an audio signal by sliding the
fader up or down. Examples of the use of faders in DAW’s are to control a track’s
volume and panning.
flange. This is an audio effect that distorts sound by applying both a short delay and
variable modulation of the frequency.
folder. A folder is a collection of related tracks and serves as a simple and convenient
means of controlling all of the tracks contained within it. For example, changes can be
made to the volume level of a folder to affect the combined volume of all of the tracks
contained within that folder. Changes made to the volume level of individual tracks
within a folder will affect the volume of that track and the relative balance between
all tracks in the folder.
frequency. The number of vibrations in a sound wave per unit of time. Frequency is
usually measured in hertz, where one hertz is one cycle per second. High-pitched sounds
have higher frequencies, and low-pitched sounds have lower frequencies.
glue. To glue a track causes some DAW’s to render the selected items to a new single WAV
file, which replaces the original items.
high-pass filter. A high-pass filter removes all sound below a set frequency. It can be
useful in removing certain kinds of rumble and hum.
input monitoring. The process of having the audio stream that is being recorded also fed
directly back to the musician or singer through his or her headphones. If latency is too
high, a perceived delay or echo effect will be noticeable.
items. Items (a.k.a. media items or media events) contain audio/MIDI information and
reside in a track. Items can be empty or can have one or more takes, one of which is
visible and “active.” Items are often called parts in other DAW software.
latency. Latency is a measurement of the time taken for audio to travel from the soundcard,
through a DAW, and out again to your headphones or speakers. Usually measured
in milliseconds. For input monitoring, a latency of less than 5 ms is usually
level. The amount of power that drives an audio signal. Common names given to varying
levels of voltage (from lowest to highest) are microphone level, instrument level, and
lossless compression. A method of compressing the size of audio files without losing any
of its frequencies. Examples of lossless compression formats are FLAC and OGG
low-pass filter. A low-pass filter removes all sound above a set frequency. It is useful in
removing certain kinds of hiss.
lossy compression. A method of compressing the size of audio files that includes stripping
out those frequencies that the human ear either doesn’t hear or hears poorly.
Examples of lossy compression are WMA and MP3.
markers. Markers, like regions, are used to identify important sections of a song (for
example, verse, chorus, intro, and so on). Unlike regions, markers indicate only the
beginning point of a section and are displayed in the Markers band of the Timeline.
Markers are useful for quickly locating and playing a section.
master. The section of a DAW’s mixer where the main mix is controlled.
media item. A media item is an audio or MIDI recording. In many DAWs, these are
known as clips.
MIDI (Musical Instrument Digital Interface). A system that uses a synthesizer to enable
your computer to play back music under program control. MIDI instructions can include
which notes to play and which instrument to simulate. MIDI can also be used to enable a
computer to control external MIDI-compliant instruments, such as synthesizers.
Mixer view. Mixer view depicts a virtual representation of a traditional hardware
mixing desk. This can be used to combine all of the signals from your various audio
and MIDI tracks into a main mix, usually consisting of one single stereo-paired output.
MP3. MPEG audio layer 3. An audio format with data compression that is widely used
to transfer music over the Internet. An MP3 file is much smaller than a WAV file, but it
still sounds to most people’s ears virtually as good as the original.
noise gate. A noise gate is a filter that detects sound levels in an audio stream and shuts
out sound when the volume falls below a determined level. It can be useful, for example,
to cut out unwanted background noise that may be present during an otherwise silent
passage on a recording.
normalization. This effect permanently adjusts the amplitude (volume, essentially) of
audio to a certain value. Normalization is particularly useful when creating a CD
using various different tracks. Normalization can be used to help obtain a consistent
level of volume for the different tracks.
notch filter. This filter removes all audio in a specified frequency band in a WAV file.
OGG Vorbis. OGG Vorbis is a free, open-source audio compression format. Though
less widely used than MP3, it can provide greater fidelity.
pan. A control that lets you determine the relative left-right balance of a mono signal
within the stereo spectrum.
plug-in. A small program used in conjunction with a DAW to enable you to manipulate
the sound of recorded tracks. Functions carried out by plug-ins include EQ, delay,
and chorus. Two widely used plug-in formats are DirectX and VST (Virtual Studio).
real-time. Refers to effects that take place as you listen, without a noticeable delay.
receive. The method by which audio sent from one track is accepted by its destination
track or bus.
regions. Regions, like markers, are used to identify important sections of a song (for
example, verse, chorus, intro, and so on). Unlike markers, regions indicate both the
beginning and ending points of a section and are displayed in the Regions band of
the Timeline. Regions are useful for relocating/duplicating existing sections of a song.
reverb. An effect that simulates natural reverberations (sound reflections) that occur in
different rooms and environments to create an ambience or sense of space.
reverse. An effect that takes one or both channels in a sound file and plays them
rip. To extract music directly from your CD in pure digital form and save it directly to
your hard drive.
routing. A term used to describe the path(s) taken by an audio stream that, on playback,
takes it from the track on which it has been recorded all the way to its position in the
Routing Matrix. Overview of a project’s entire routing network. Changes to
routing can also be made in the Routing Matrix.
sampler. A hardware device or software application that uses samples to generate audio
output. Samplers often use a number of samples together to create realistic-sounding
reproductions of real sounds and musical instruments.
sampling rate. The sampling rate is a measurement of how many times a second your
audio recording software captures the incoming audio signal. Commonly used sampling
rates are 44.1 kHz, 48 kHz, 88.2 kHz, and 96 kHz. For example, at 44.1 kHz, incoming
audio is sampled 44,100 times per second. The higher the sampling rate, the more disk
storage required. If you’re not sure which rate to use, try recording initially with a sampling
rate of 44.1 and see whether you are satisfied with the results.
send. The method by which audio is routed from one track to another track or bus,
where it is accepted via a receive.
stems. Stems are a group of selected tracks and/or folders. They allow you to render
selected tracks to disk at the same time as (or in lieu of) the main mix. You can use
stems to “freeze” FX on tracks or to render each track in a mix (a stem, as it is called) so
that a mastering engineer (or whoever) can later adjust the mix.
synthesizer. A hardware or software device that artificially (using oscillators) generates
signals to simulate the sounds of real instruments or to create other sounds not possible
with real instruments.
takes. A take is a part of an item that contains a media source (audio, MIDI, or other
type). An item can have multiple takes, which may refer to different recorded versions
(where it gets its name). When you adjust an item (splitting, adjusting its start/end times,
stretching, and so on), the action is performed on all takes, so that if you need to switch
to a different take, the overall timing is correct.
tracks. A project is made up of any number of tracks that you record (or import). Tracks can contain multiple media items and envelopes. Tracks appear as horizontal bars on the Timeline and as vertical bars on the mixer. There can be three types of tracks: 1) standard tracks, which hold items and envelopes; 2) folders, which hold items, envelopes, and standard tracks; and 3) stems, which include selected standard tracks/folders.
Track Control Panel (TCP). The area where a project’s various tracks are listed. So called because it is used to control the various track attributes, such as to arm for recording or to control volume on playback.
VST and VSTi. A widely used plug-in format, Virtual Studio Technology, developed
originally by Steinberg. The term VST usually refers to FX that manipulate or modify
the sound of audio in some way. VSTi refers to a virtual instrument used for sound
wave (.wav) files. A wave file is an uncompressed audio file on your PC. This enables it
to be as close a copy to the original analog data as is possible.
waveform. A visual representation of a WAV file.
WAV editor. Audio software designed for editing digital audio.
wet. A term used to describe an audio signal to which effects have been added. The
opposite is dry.
That raps up todays blog I think. Please feel free to add to this list and also to give any clearer definitions of the above. I'm sure over the next month or so with all your help, we can put together the best technical term glossary for all of us to refer to.