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These terms are used in many different audio contexts.  Understanding them is important to knowing how to operate audio equipment in general, whether computer-based or not.
These terms are used in many different audio contexts.  Understanding them is important to knowing how to operate audio equipment in general, whether computer-based or not.
 
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=== MIDI Sequencer ===
=== MIDI Sequencer ===
A '''sequencer''' is a device or software application which produces signals that a synthesizer to turns into sound.  Sequencers also allow the editing and arrangement of these signals into a musical composition.  The two MIDI-focussed DAWs in the Musicians' Guide, Qtractor and Rosegarden, are well-suited to serve as MIDI sequencersFurthermore, all three DAWs use MIDI instructions to perform automation.
A '''sequencer''' is a device or software program that produces signals that a synthesizer turns into sound.  You can also use a sequencer to arrange MIDI signals into music.  The Musicians' Guide covers two digital audio workstations (DAWs) that are primarily MIDI sequencers, Qtractor and Rosegarden.  All three DAWs in this guide use MIDI signals to control other devices or effects.


=== Busses, Master Bus, and Sub-master Bus ===
=== Busses, Master Bus, and Sub-master Bus ===
An '''audio bus''' is used to send an audio signal from one place to another.  Like a highway is used to collect all sorts of automobiles going in the same general direction, an audio bus is collects all sorts of signals going in the same direction (usually into or out of the DAW, but sometimes between parts of the DAW).  Unlike a highway, however, once audio enters a bus, it cannot be separated.
All audio being routed out of a program usually passes through the master bus.  The '''master bus''' collects and consolidates all audio and MIDI tracks, allowing for final level adjustments and for simpler mastering.  The primary purpose of the master bus is to mix all of the tracks into two channels.
A '''sub-master bus''' collects audio signals before they're inputted to the master bus.  Using a sub-master bus is optional, and it allows you to subject more than one track to the same adjustment, without affecting all the tracks.  A sub-master bus is often simply referred to as a bus.
<!-- [[File:FMG-bus.xcf]] -->
<!-- [[File:FMG-bus.xcf]] -->
<!-- [[File:FMG-master_sub_bus.xcf]] -->
<!-- [[File:FMG-master_sub_bus.xcf]] -->
[[File:FMG-bus.png|200px|How audio busses work.]]
[[File:FMG-bus.png|200px|How audio busses work.]]
[[File:FMG-master_sub_bus.png|200px|The relationship between the master bus and sub-master busses.]]
[[File:FMG-master_sub_bus.png|200px|The relationship between the master bus and sub-master busses.]]
An '''audio bus''' sends audio signals from one place to another.  Many different signals can be inputted to a bus simultaneously, and many different devices or applications can read from a bus simultaneously.  Signals inputted to a bus are mixed together, and cannot be separated after entering a bus.  All devices or applications reading from a bus receive the same signal.
All audio routed out of a program passes through the master bus.  The '''master bus''' combines all audio tracks, allowing for final level adjustments and simpler mastering.  The primary purpose of the master bus is to mix all of the tracks into two channels.
A '''sub-master bus''' combines audio signals before they reach the master bus.  Using a sub-master bus is optional.  They allow you to adjust more than one track in the same way, without affecting all the tracks.
Audio busses are also used to send audio into effects processors.


=== Level (Volume/Loudness) ===
=== Level (Volume/Loudness) ===
The perceived '''volume''' or '''loudness''' of a portion of audio is a complex phenomenon, involving many different factors that are too numerous and enigmatic to explain anywhere, let alone here.  One widely-agreed method of assessing loudness is by measuring the sound pressure level (SPL), which is measured in Bels or decibels (B or dB).  In audio production communities, this is normally referred to simply as "level."  The '''level''' of an audio signal is one way of measuring the signal's perceived loudness, and it is stored along with the audio signal.
The perceived '''volume''' or '''loudness''' of sound is a complex phenomenon, not entirely understood by experts.  One widely-agreed method of assessing loudness is by measuring the sound pressure level (SPL), which is measured in decibels (dB) or bels (B, equal to ten decibels).  In audio production communities, this is called "level."  The '''level''' of an audio signal is one way of measuring the signal's perceived loudness.  The level is part of the information stored in an audio file.


Much controversy exists over how to effectively monitor and adjust levels, partly because it is an aesthetic practice and therefore heavily subjectiveCommonly, the average level is designed to be -6dB on the meter, and the maximum level 0dB, but this is not the case with all metering practices.
There are many different ways to monitor and adjust the level of an audio signal, and there is no widely-agreed practice.  One reason for this situation is the technical limitations of recorded audio.  Most level meters are designed so that the average level is -6&nbsp;dB on the meter, and the maximum level is 0&nbsp;dB.  This practice was developed for analog audio.  We recommend using an external meter and the "K-system," described in a link below.  The K-system for level metering was developed for digital audio.


Throughout the Fedora Musicians' Guide, this term is often called "volume level," to avoid confusion with other levels, and to be clear that this is not referring to perceived volume or loudness.
In the Musicians' Guide, this term is called "volume level," to avoid confusion with other levels, or with perceived volume or loudness.


For more information, refer to these web pages:
For more information, refer to these web pages:
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=== Panning and Balance ===
=== Panning and Balance ===
'''Panning''' allows you to adjust the portion of a channel's signal is sent through each output channel. Assuming a stereophonic (two-channel) setup, the two channels will represent the "left" and the "right" speakers.  The DAW will have two channels of audio recorded, and a default setup might send all of the "left" recorded channel to the "left" output channel, and all of the "right" recorded channel to the "right" output channel. Panning allows you to adjust this, sending some of the left recorded channel's volume to the right output channel, for example.  It's important to recognize that this relies on each recorded channel having a constant total output level, which is divided between the two output channels.
[[File:FMG-Balance_and_Panning.png|200px|left|The difference between adjusting panning and adjusting balance.]]
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An audio engineer might initially set the left recorded channel to "full left" panning, which outputs 100% of the output level to the left output channel, and 0% to the right output channelIf they found this setup too tiring for the listener, they might adjust the left recorded channel to output 80% of the signal to the left output channel and 20% of the signal to the right output channel.  If they wanted to make the left recorded channel sound like it's in front of the listener, they could set it to "centre" panning, with 50% of the output level being outputted to each output channel.
'''Panning''' adjusts the portion of a channel's signal that is sent to each output channel.  In a stereophonic (two-channel) setup, the two channels represent the "left" and the "right" speakersTwo channels of recorded audio are available in the DAW, and the default setup sends all of the "left" recorded channel to the "left" output channel, and all of the "right" recorded channel to the "right" output channel.  Panning sends some of the left recorded channel's level to the right output channel, or some of the right recorded channel's level to the left output channel.  Each recorded channel has a constant total output level, which is divided between the two output channels.


Balance seems to have a similar effect, but it is not the same as panning.  The two terms are sometimes confused on audio equipment and in popular usage.  Adjusting the '''balance''' allows you to change the volume of the output channels, without redirecting the recorded signal.  The default setting for balance should be at "centre," meaning 0% deviation to left or right.  As the dial is adjusted towards the "full left" setting, the volume of the left recorded channel and the left output channel remain unchanged, but the volume level of the right output channel is decreasedAs the dial is adjusted towards the "full right" setting, the volume of the right recorded channel and the right output channel remain unchanged, but the volume level of the left output channel is decreasedIf the dial is adjusted to "20% left," the audio equipment would reduce the volume level of the right channel by 20%, increasing the perceived loudness of the left channel by approximately 20%.
The default setup for a left recorded channel is for "full left" panning, meaning that 100% of the output level is output to the left output channel.  An audio engineer might adjust this so that 80% of the recorded channel's level is output to the left output channel, and 20% of the level is output to the right output channel.  An audio engineer might make the left recorded channel sound like it is in front of the listener by setting the panner to "center," meaning that 50% of the output level is output to both the left and right output channels.


The balance is not generally adjusted until final play-back.  The effect is intended to be used to compensate for poorly set-up listening environments, where the speakers are not equal distances from the listnerIf the left speaker is closer to the listener than the right speaker, for example, the listener could adjust the balance to the right, so that the volume level of the left speaker would be lowerWhen the listener does this, they allow an equal perceived loudness to reach their ears from both speakers.  For many reasons, this solution is not ideal: it is better to have properly set-up speakers, but sometimes it is also impossible or impractical.
Balance is sometimes confused with panning, even on commercially-available audio equipment.  Adjusting the '''balance''' changes the volume level of the output channels, without redirecting the recorded signal.  The default setting for balance is "center," meaning 0% change to the volume levelAs you adjust the dial from "center" toward the "full left" setting, the volume level of the right output channel is decreased, and the volume level of the left output channel remains constant.  As you adjust the dial from "center" toward the "full right" setting, the volume level of the left output channel is decreased, and the volume level of the right output channel remains constantIf you set the dial to "20% left," the audio equipment would reduce the volume level of the right output channel by 20%, increasing the perceived loudness of the left output channel by approximately 20%.


[[File:FMG-Balance_and_Panning.png|200px|left|The difference between adjusting panning and adjusting balance.]]
You should adjust the balance so that you perceive both speakers as equally loud. Balance compensates for poorly set up listening environments, where the speakers are not equal distances from the listener.  If the left speaker is closer to you than the right speaker, you can adjust the balance to the right, which decreases the volume level of the left speaker. This is not an ideal solution, but sometimes it is impossible or impractical to set up your speakers correctly.  You should adjust the balance only at final playback.
<!-- [[File:FMG-Balance_and_Panning.xcf]] -->


=== Time, Timeline and Time-Shifting ===
=== Time, Timeline and Time-Shifting ===
There are many ways to measure musical time.  The four most popular scales for digital audio are:
There are many ways to measure musical time.  The four most popular time scales for digital audio are:
* Bars and Beats: Usually used for MIDI work, and called "BBT," meaning "Bars, Beats, and Ticks."  A tick is a partial beat.
* Bars and Beats: Usually used for MIDI work, and called "BBT," meaning "Bars, Beats, and Ticks."  A tick is a partial beat.
* Minutes and Seconds: Usually used for audio work.
* Minutes and Seconds: Usually used for audio work.
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* Samples: Relating directly to the format of the underlying audio file, a sample is the shortest possible length of time in an audio file.  See [[User:Crantila/FSC/Sound_Cards#Sample_Rate|this section]] for more information on samples.
* Samples: Relating directly to the format of the underlying audio file, a sample is the shortest possible length of time in an audio file.  See [[User:Crantila/FSC/Sound_Cards#Sample_Rate|this section]] for more information on samples.


Most audio software, particularly Digital Audio Workstations (DAWs), allow the user to choose which scale they prefer.  They use a '''timeline''' to display the progression of time in a session or file, allowing the user to do something called '''time-shifting''', which is adjusting the point in the timeline that a region starts to be played.
Most audio software, particularly digital audio workstations (DAWs), allow the user to choose which scale they prefer.  DAWs use a '''timeline''' to display the progression of time in a session, allowing you to do '''time-shifting'''; that is, adjust the time in the timeline when a region starts to be played.


Time is usually represented horizontally, with the leftmost point being the beginning of the session (zero, regardless of the unit of measurement), and the rightmost point being some distance after the end of the session.
Time is represented horizontally, where the leftmost point is the beginning of the session (zero, regardless of the unit of measurement), and the rightmost point is some distance after the end of the session.


=== Synchronization ===
=== Synchronization ===
'''Synchronization''' is exactly what it sounds like - synchronizing the operation of multiple tools.  Most often this is used to synchronize movement of the transport, and to control automation across applications and devices.  This sort of synchronization is typically achieved with MIDI channels that are not used directly for synthesis.
'''Synchronization''' is synchronizing the operation of multiple tools, frequently the movement of the transport.  Synchronization also controls automation across applications and devices.  MIDI signals are usually used for synchronization.


=== Routing and Multiplexing ===
=== Routing and Multiplexing ===
'''Routing''' audio is transmitting a signal from one place to another - between applications, between parts of applications, or between devices.  On GNU/Linux systems, audio routing in audio creation applications is normally achieved with the JACK Audio Connection Kit.  JACK-aware applications (and PulseAudio ones, if so configured) provide the JACK server with different inputs and outputs depending on the current application's configuration.  Most applications connect themselves in a default arrangement, but they can always be re-configured with the QjackCtl graphical interface, and often within other programs. This allows for maximum flexibility and creative solutions to otherwise-complex problems: you can easily re-route the output of a program like FluidSynth so that it is provided to Ardour, for recording.
[[File:FMG-routing_and_multiplexing.png|200px|left|Illustration of routing and multiplexing in the "Connections" window of the QjackCtl interface.]]
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'''Multiplexing''' is a related concept, which allows the connection of multiple inputs and outputs to a single connectionThis may not seem like an important or difficult thing to do, but remember that digital, computer-based audio evolved from analogue audio, where a special "multiplexer" device is required to connect multiple devices to one connection.
'''Routing''' audio transmits a signal from one place to another - between applications, between parts of applications, or between devices.  On Linux systems, the JACK Audio Connection Kit is used for audio routing.  JACK-aware applications (and PulseAudio ones, if so configured) provide inputs and outputs to the JACK server, depending on their configuration.  The QjackCtl application can adjust the default connectionsYou can easily reroute the output of a program like FluidSynth so that it can be recorded by Ardour, for example, by using QjackCtl.


[[File:FMG-routing_and_multiplexing.png|200px|left|Illustration of routing and multiplexing in the "Connections" window of the QjackCtl interface.]]
'''Multiplexing''' allows you to connect multiple devices and applications to a single input or output. QjackCtl allows you to easily perform multiplexing.  This may not seem important, but remember that only one connection is possible with a physical device like an audio interface. Before computers were used for music production, multiplexing required physical devices to split or combine the signals.
<!-- [[FMG-routing_and_multiplexing.xcf]] -->


=== Multichannel Audio ===
=== Multichannel Audio ===
An '''audio channel''' is a single path for delivering audio data.  '''Multichannel audio''' is any audio which uses more than one channel simultaneously, allowing the storage and transmission of more audio data than single-channel audio.
An '''audio channel''' is a single path of audio data.  '''Multichannel audio''' is any audio which uses more than one channel simultaneously, allowing the transmission of more audio data than single-channel audio.


Audio was originally recorded with only one channel, producing "monophonic," or "mono" recordings.  Beginning in the 1950s, stereophonic recordings, with two independent channels, gradually began replacing monophonic recordings.  Since humans have two independent ears, it makes a certain amount of sense to record and reproduce audio with two independent channels, involving two speakers.  Most sound recordings available today are stereophonic, and people have found this mostly satisfying.
Audio was originally recorded with only one channel, producing "monophonic," or "mono" recordings.  Beginning in the 1950s, stereophonic recordings, with two independent channels, began replacing monophonic recordings.  Since humans have two independent ears, it makes sense to record and reproduce audio with two independent channels, involving two speakers.  Most sound recordings available today are stereophonic, and people have found this mostly satisfying.


There is a growing trend, however, towards five- and seven-channel audio, driven primarily by "surround-sound" movies, and not widely available for music.  Two transmission formats do exist for music, DVD Audio (DVD-A) and Super Audio CD (SACD).  The development of these formats, and the devices to use them, is held back by the proliferation of headphones with personal MP3 players, a general lack of desire for improvement in audio quality amongst consumers, and the copy-protection measures put in place by record labels.  The result is that, while some consumers are willing to pay moderately higher prices for these formats, there is a poor selection of recordings available.  Even if somebody were to buy a DVD-A or SACD-capable player, they would need to replace the rest of their audio equipment with models that support proprietary copy-protection software, or the player is often forbidden from outputting audio signals at a higher quality than available from a conventional Audio CD.  None of these factors, unfortunately, seems about to change in the near future.
There is a growing trend toward five- and seven-channel audio, driven primarily by "surround-sound" movies, and not widely available for music.  Two "surround-sound" formats exist for music: DVD Audio (DVD-A) and Super Audio CD (SACD).  The development of these formats, and the devices to use them, is held back by the proliferation of headphones with personal MP3 players, a general lack of desire for improvement in audio quality amongst consumers, and the copy-protection measures put in place by record labels.  The result is that, while some consumers are willing to pay higher prices for DVD-A or SACD recordings, only a small number of recordings are available.  Even if you buy a DVD-A or SACD-capable player, you would need to replace all of your audio equipment with models that support proprietary copy-protection software.  Without this equipment, the player is often forbidden from outputting audio with a higher sample rate or sample format than a conventional audio CD.  None of these factors, unfortunately, seem like they will change in the near future.

Latest revision as of 05:55, 2 August 2010

These terms are used in many different audio contexts. Understanding them is important to knowing how to operate audio equipment in general, whether computer-based or not.

MIDI Sequencer

A sequencer is a device or software program that produces signals that a synthesizer turns into sound. You can also use a sequencer to arrange MIDI signals into music. The Musicians' Guide covers two digital audio workstations (DAWs) that are primarily MIDI sequencers, Qtractor and Rosegarden. All three DAWs in this guide use MIDI signals to control other devices or effects.

Busses, Master Bus, and Sub-master Bus

How audio busses work. The relationship between the master bus and sub-master busses.

An audio bus sends audio signals from one place to another. Many different signals can be inputted to a bus simultaneously, and many different devices or applications can read from a bus simultaneously. Signals inputted to a bus are mixed together, and cannot be separated after entering a bus. All devices or applications reading from a bus receive the same signal.

All audio routed out of a program passes through the master bus. The master bus combines all audio tracks, allowing for final level adjustments and simpler mastering. The primary purpose of the master bus is to mix all of the tracks into two channels.

A sub-master bus combines audio signals before they reach the master bus. Using a sub-master bus is optional. They allow you to adjust more than one track in the same way, without affecting all the tracks.

Audio busses are also used to send audio into effects processors.

Level (Volume/Loudness)

The perceived volume or loudness of sound is a complex phenomenon, not entirely understood by experts. One widely-agreed method of assessing loudness is by measuring the sound pressure level (SPL), which is measured in decibels (dB) or bels (B, equal to ten decibels). In audio production communities, this is called "level." The level of an audio signal is one way of measuring the signal's perceived loudness. The level is part of the information stored in an audio file.

There are many different ways to monitor and adjust the level of an audio signal, and there is no widely-agreed practice. One reason for this situation is the technical limitations of recorded audio. Most level meters are designed so that the average level is -6 dB on the meter, and the maximum level is 0 dB. This practice was developed for analog audio. We recommend using an external meter and the "K-system," described in a link below. The K-system for level metering was developed for digital audio.

In the Musicians' Guide, this term is called "volume level," to avoid confusion with other levels, or with perceived volume or loudness.

For more information, refer to these web pages:

Panning and Balance

The difference between adjusting panning and adjusting balance.
The difference between adjusting panning and adjusting balance.

Panning adjusts the portion of a channel's signal that is sent to each output channel. In a stereophonic (two-channel) setup, the two channels represent the "left" and the "right" speakers. Two channels of recorded audio are available in the DAW, and the default setup sends all of the "left" recorded channel to the "left" output channel, and all of the "right" recorded channel to the "right" output channel. Panning sends some of the left recorded channel's level to the right output channel, or some of the right recorded channel's level to the left output channel. Each recorded channel has a constant total output level, which is divided between the two output channels.

The default setup for a left recorded channel is for "full left" panning, meaning that 100% of the output level is output to the left output channel. An audio engineer might adjust this so that 80% of the recorded channel's level is output to the left output channel, and 20% of the level is output to the right output channel. An audio engineer might make the left recorded channel sound like it is in front of the listener by setting the panner to "center," meaning that 50% of the output level is output to both the left and right output channels.

Balance is sometimes confused with panning, even on commercially-available audio equipment. Adjusting the balance changes the volume level of the output channels, without redirecting the recorded signal. The default setting for balance is "center," meaning 0% change to the volume level. As you adjust the dial from "center" toward the "full left" setting, the volume level of the right output channel is decreased, and the volume level of the left output channel remains constant. As you adjust the dial from "center" toward the "full right" setting, the volume level of the left output channel is decreased, and the volume level of the right output channel remains constant. If you set the dial to "20% left," the audio equipment would reduce the volume level of the right output channel by 20%, increasing the perceived loudness of the left output channel by approximately 20%.

You should adjust the balance so that you perceive both speakers as equally loud. Balance compensates for poorly set up listening environments, where the speakers are not equal distances from the listener. If the left speaker is closer to you than the right speaker, you can adjust the balance to the right, which decreases the volume level of the left speaker. This is not an ideal solution, but sometimes it is impossible or impractical to set up your speakers correctly. You should adjust the balance only at final playback.

Time, Timeline and Time-Shifting

There are many ways to measure musical time. The four most popular time scales for digital audio are:

  • Bars and Beats: Usually used for MIDI work, and called "BBT," meaning "Bars, Beats, and Ticks." A tick is a partial beat.
  • Minutes and Seconds: Usually used for audio work.
  • SMPTE Timecode: Invented for high-precision coordination of audio and video, but can be used with audio alone.
  • Samples: Relating directly to the format of the underlying audio file, a sample is the shortest possible length of time in an audio file. See this section for more information on samples.

Most audio software, particularly digital audio workstations (DAWs), allow the user to choose which scale they prefer. DAWs use a timeline to display the progression of time in a session, allowing you to do time-shifting; that is, adjust the time in the timeline when a region starts to be played.

Time is represented horizontally, where the leftmost point is the beginning of the session (zero, regardless of the unit of measurement), and the rightmost point is some distance after the end of the session.

Synchronization

Synchronization is synchronizing the operation of multiple tools, frequently the movement of the transport. Synchronization also controls automation across applications and devices. MIDI signals are usually used for synchronization.

Routing and Multiplexing

Illustration of routing and multiplexing in the "Connections" window of the QjackCtl interface.
Illustration of routing and multiplexing in the "Connections" window of the QjackCtl interface.

Routing audio transmits a signal from one place to another - between applications, between parts of applications, or between devices. On Linux systems, the JACK Audio Connection Kit is used for audio routing. JACK-aware applications (and PulseAudio ones, if so configured) provide inputs and outputs to the JACK server, depending on their configuration. The QjackCtl application can adjust the default connections. You can easily reroute the output of a program like FluidSynth so that it can be recorded by Ardour, for example, by using QjackCtl.

Multiplexing allows you to connect multiple devices and applications to a single input or output. QjackCtl allows you to easily perform multiplexing. This may not seem important, but remember that only one connection is possible with a physical device like an audio interface. Before computers were used for music production, multiplexing required physical devices to split or combine the signals.

Multichannel Audio

An audio channel is a single path of audio data. Multichannel audio is any audio which uses more than one channel simultaneously, allowing the transmission of more audio data than single-channel audio.

Audio was originally recorded with only one channel, producing "monophonic," or "mono" recordings. Beginning in the 1950s, stereophonic recordings, with two independent channels, began replacing monophonic recordings. Since humans have two independent ears, it makes sense to record and reproduce audio with two independent channels, involving two speakers. Most sound recordings available today are stereophonic, and people have found this mostly satisfying.

There is a growing trend toward five- and seven-channel audio, driven primarily by "surround-sound" movies, and not widely available for music. Two "surround-sound" formats exist for music: DVD Audio (DVD-A) and Super Audio CD (SACD). The development of these formats, and the devices to use them, is held back by the proliferation of headphones with personal MP3 players, a general lack of desire for improvement in audio quality amongst consumers, and the copy-protection measures put in place by record labels. The result is that, while some consumers are willing to pay higher prices for DVD-A or SACD recordings, only a small number of recordings are available. Even if you buy a DVD-A or SACD-capable player, you would need to replace all of your audio equipment with models that support proprietary copy-protection software. Without this equipment, the player is often forbidden from outputting audio with a higher sample rate or sample format than a conventional audio CD. None of these factors, unfortunately, seem like they will change in the near future.