Wednesday, 30 March 2016
6 Tips For Balancing The Bass And Drum Mix
6 Tips For Balancing The Bass And Drum Mix
Perhaps the most difficult task of a mixing engineer is balancing the bass and drums (especially the bass and kick). Nothing can make or break a mix faster than the way these instruments work together. It’s not uncommon for a mixer to spend hours on this balance (both level and frequency) because if the relationship isn’t correct, then the song will just never sound big and punchy.
So how do you get this mysterious balance?
In order to have the impact and punch that most modern mixes exhibit, you have to make a space in your mix for both of these instruments so they won't fight each other and turn into a muddy mess. While simply EQing your bass high and your kick low (or the other way around), might work at it’s simplest, it’s best to have a more in-depth strategy, so consider the following:
1) EQ the kick drum between 60 to120Hz as this will allow it to be heard on smaller speakers. For more attack and beater click add between 1k to 4kHz. You may also want to dip some of the boxiness between 200-500Hz. EQing in the 30-60Hz range will produce a kick that you can feel, but it may also sound thin on smaller speakers and probably won’t translate well to a variety of speaker systems. Most 22" kick drums are centered somewhere around 80Hz anyway.
2) Bring up the bass with the kick. The kick and bass should occupy slightly different frequency spaces. The kick will usually be in the 60 to 80Hz range whereas the bass will emphasize higher frequencies anywhere from 80 to 250Hz (although sometimes the two are reversed depending upon the song). Shelve out any unnecessary bass frequencies (below 30Hz on kick and below 50Hz on the bass, although the frequency for both may be as high as 60Hz according to style of the song and your taste) so they're not boomy or muddy. There should be a driving, foundational quality to the combination of these two together.
A common mistake is to emphasize the kick with either too much level or EQ, while not featuring enough of the bass guitar (see the graphic on the left for a good visual of what it sounds like). This gives you the illusion that your mix is bottom light, because what you’re doing is shortening the duration of the low frequency envelope in your mix. Since the kick tends to be more transient than the bass guitar, this gives you the idea that the low frequency content of your mix is inconsistent. For Pop music, it is best to have the kick provide the percussive nature of the bottom while the bass fills out the sustain and musical parts.
3) Make sure that the snare is strong, otherwise the song will lose its drive when the other instruments are added in. This usually calls for at least some compression, especially if the snare hits are inconsistent throughout the song. You may need a small EQ boost at 1kHz for attack, 120 to 240Hz for fullness, and 10k for snap. As you bring in the other drums and cymbals, you might want to dip a little of 1kHz on these to make room for the snare. Also make sure that the toms aren't too boomy (if so, shelve out the frequencies below 60 Hz).
4) If you’re having trouble with the mix because it's sounding cloudy and muddy on the bottom end, mute both the kick drum and bass to determine what else might be in the way in the low end. You might not realize that there are some frequencies in the mix that aren't really musically necessary. With piano or guitar, you're mainly looking for the mids and top end to cut through, while the low-end is just getting in the way, so it’s best to clear some of that out with a hi-pass filter. When soloed, the instrument might sound too thin, but with the rest of the mix the low-end will now sound so much better and you won’t be missing that low end from the other instruments. Now the mix sounds louder, clearer, and fuller. Be careful not to cut too much from the other instruments, as you might loose the warmth of the mix.
5) For Dance music, be aware of kick drum to bass melody dissonance. The bass line over the huge sound systems in today's clubs is very important and needs to work very well with the kick drum. But if your kick is centered around an A note and the bass line is tuned to A#, it's going to clash. Tune your kick samples to the bass lines (or vice versa) where needed.
6) If you feel that you don't have enough bass or kick, boost the level, not the EQ. This is a mistake that everyone makes when their first getting their mixing chops together. Most bass drums and bass guitars have plenty of low end and don't need much more, so be sure that their level together and with the rest of the mix is correct before you go adding EQ. Even then, a little goes a long way.
Perhaps the most difficult task of a mixing engineer is balancing the bass and drums (especially the bass and kick). Nothing can make or break a mix faster than the way these instruments work together. It’s not uncommon for a mixer to spend hours on this balance (both level and frequency) because if the relationship isn’t correct, then the song will just never sound big and punchy.
So how do you get this mysterious balance?
In order to have the impact and punch that most modern mixes exhibit, you have to make a space in your mix for both of these instruments so they won't fight each other and turn into a muddy mess. While simply EQing your bass high and your kick low (or the other way around), might work at it’s simplest, it’s best to have a more in-depth strategy, so consider the following:
1) EQ the kick drum between 60 to120Hz as this will allow it to be heard on smaller speakers. For more attack and beater click add between 1k to 4kHz. You may also want to dip some of the boxiness between 200-500Hz. EQing in the 30-60Hz range will produce a kick that you can feel, but it may also sound thin on smaller speakers and probably won’t translate well to a variety of speaker systems. Most 22" kick drums are centered somewhere around 80Hz anyway.
2) Bring up the bass with the kick. The kick and bass should occupy slightly different frequency spaces. The kick will usually be in the 60 to 80Hz range whereas the bass will emphasize higher frequencies anywhere from 80 to 250Hz (although sometimes the two are reversed depending upon the song). Shelve out any unnecessary bass frequencies (below 30Hz on kick and below 50Hz on the bass, although the frequency for both may be as high as 60Hz according to style of the song and your taste) so they're not boomy or muddy. There should be a driving, foundational quality to the combination of these two together.
A common mistake is to emphasize the kick with either too much level or EQ, while not featuring enough of the bass guitar (see the graphic on the left for a good visual of what it sounds like). This gives you the illusion that your mix is bottom light, because what you’re doing is shortening the duration of the low frequency envelope in your mix. Since the kick tends to be more transient than the bass guitar, this gives you the idea that the low frequency content of your mix is inconsistent. For Pop music, it is best to have the kick provide the percussive nature of the bottom while the bass fills out the sustain and musical parts.
3) Make sure that the snare is strong, otherwise the song will lose its drive when the other instruments are added in. This usually calls for at least some compression, especially if the snare hits are inconsistent throughout the song. You may need a small EQ boost at 1kHz for attack, 120 to 240Hz for fullness, and 10k for snap. As you bring in the other drums and cymbals, you might want to dip a little of 1kHz on these to make room for the snare. Also make sure that the toms aren't too boomy (if so, shelve out the frequencies below 60 Hz).
4) If you’re having trouble with the mix because it's sounding cloudy and muddy on the bottom end, mute both the kick drum and bass to determine what else might be in the way in the low end. You might not realize that there are some frequencies in the mix that aren't really musically necessary. With piano or guitar, you're mainly looking for the mids and top end to cut through, while the low-end is just getting in the way, so it’s best to clear some of that out with a hi-pass filter. When soloed, the instrument might sound too thin, but with the rest of the mix the low-end will now sound so much better and you won’t be missing that low end from the other instruments. Now the mix sounds louder, clearer, and fuller. Be careful not to cut too much from the other instruments, as you might loose the warmth of the mix.
5) For Dance music, be aware of kick drum to bass melody dissonance. The bass line over the huge sound systems in today's clubs is very important and needs to work very well with the kick drum. But if your kick is centered around an A note and the bass line is tuned to A#, it's going to clash. Tune your kick samples to the bass lines (or vice versa) where needed.
6) If you feel that you don't have enough bass or kick, boost the level, not the EQ. This is a mistake that everyone makes when their first getting their mixing chops together. Most bass drums and bass guitars have plenty of low end and don't need much more, so be sure that their level together and with the rest of the mix is correct before you go adding EQ. Even then, a little goes a long way.
Thursday, 17 March 2016
That church sound can be better ......
or distribute it to a wider audience.
Monday, 4 January 2016
Hints and Tips when Recording:
can avoid the expense of buying a separate amp to create a
headphone mix. Plug your headphones into the console’s
headphone connector and use its monitor mix for your
foldback. Alter channel fader levels as you wish to achieve
optimum headphone levels for your performance.
• If your console is not large enough to cope with every
multitrack send and return, connect only as many Direct
Outs as you need per take. For example, if you are
recording solo you will only be recording one instrument at
a time anyway, so a maximum of only two direct outs will
be required for stereo instruments, and one for mono ones.
The same channel direct outs may then be repatched to
adjacent multitrack tape ins to record new tracks. This
should leave enough channels free to monitor all your
recorded tracks.
• If you run out of tape tracks, group instruments together.
For example a fully mic’d up drumkit can be recorded in
stereo to two tape tracks via a pair of groups, or if you are
really stretched you could do this with the entire rhythm
section, including bass and rhythm guitar. However, it is
then essential to mix the balance between the instruments
accurately as, once recorded, they can never be individually
altered again.
• If you have only one effects unit and you need it to create a
variety of different sounds, it may be neccessary to record
the instrument with effects included. Again, remember that
once you have done this there is no going back, so wherever
possible it is best to record “dry” and buy a second effects
unit if you can.
X32 MULTI-TRACK SYSTEM
Monitor Mixing
between the FOH and monitor engineer so that they can
talk to each other during the performance.
• Each stage monitor needs its own power amp. Keep
things tidy by using rack-mounted stereo amps.
• Graphic EQs are patched via the console, like the power
amps they should be rack-mounted for easy access.
• If the lead vocalist uses in-ear monitoring, he/she will be
acoustically isolated, so it’s a good idea to feed audience
pick-up mics into his/her mix to provide a sense of
involvement.
• ‘Side fills’ are often used where monitoring is required
over a large stage area, floor space is at a premium, and
too many wedge monitors would simply clutter things up
both physically and acoustically. Don’t compromise on
these speakers - they’ll have to work hard to punch sound
through to the performers.
• The Monitor Engineer’s wedge lets him hear the total
foldback mix or selected parts thereof.
• A good Monitor Engineer, who is “invisible” to the
audience, will always position himself so as to see visual
signals from the performers.
Monitor Mixing
themselves.
When dealing with the monitoring requirements of, say,
a large live band, it is common practice to keep the
monitor mix function totally separate from the Front of
House console.
Some form of graphic equaliser in line with each monitor
speaker is desirable as it allows troublesome frequencies to
be notched out. The monitor system is rung out in exactly
the same way as the main PA and the final ringing out must be done with both the
monitor and main PA systems set at their normal operating
level.The monitoring console is situated off-stage and
derives its feed direct from mic splitters. Note: the Spirit
Monitor 2 console has its own built-in mic splitters.
Recording Live
performance to avoid distraction from the live sound.
If this is not possible, use a good pair of noise-excluding
headphones for monitoring.
• Wherever possible, take feeds from mic splitters - this
will provide clean, low-noise signals suitable for
recording.
• Often, Tape Sends are unbalanced, so keep signal paths as
short as possible between output and recorder to avoid
interference.
• If there aren’t enough microphones, use a stereo pair to
pick up the overall sound and the rest to emphasize
individual performers.
• Use a compressor/limiter to avoid overloading the digital
input of the recorder.
Recording Live
In some situations, you may want to record a performance.
Depending on the situation, the feed for recording may
come from the FOH mixer, microphone splitter boxes, or
your own microphones which have been set up alongside
those of the band.
The diagram below shows a typical example of the sound
sources being split between FOH and Recording. The
recording console operates independently from the FOH
mixer.
Cables and Connections
Cables and Connections
Interference and hum can be avoided! A few minutes spent
checking cable runs and connectors pays dividends.
• A balanced audio connection provides low noise operation
by cancelling out any interference in a signal. It does this
by using a 2-conductor mic cable surrounded by a shield.
Any interference picked up will be of the same polarity on
the two conductors and is therefore rejected by the mic
input’s Differential Amplifier.
• Don’t skimp on interconnecting cables - always buy the
best that you can afford. Make sure that all connections are
sound and keep cable runs as short as is practicable.
• A multicore cable and stage box will keep trailing cables to
a minimum and presents a tidy and practical approach.
• If your mixer has a separate power supply unit, keep it well
away from the console.
• Where signal and mains cables must cross, make sure
they’re at 90° to each other. This will help reduce the risk
of hum and noise.
• If the venue has a three-phase supply, don’t share the same
phase as lighting controllers.
Interference and hum can be avoided! A few minutes spent
checking cable runs and connectors pays dividends.
• A balanced audio connection provides low noise operation
by cancelling out any interference in a signal. It does this
by using a 2-conductor mic cable surrounded by a shield.
Any interference picked up will be of the same polarity on
the two conductors and is therefore rejected by the mic
input’s Differential Amplifier.
• Don’t skimp on interconnecting cables - always buy the
best that you can afford. Make sure that all connections are
sound and keep cable runs as short as is practicable.
• A multicore cable and stage box will keep trailing cables to
a minimum and presents a tidy and practical approach.
• If your mixer has a separate power supply unit, keep it well
away from the console.
• Where signal and mains cables must cross, make sure
they’re at 90° to each other. This will help reduce the risk
of hum and noise.
• If the venue has a three-phase supply, don’t share the same
phase as lighting controllers.
Tuesday, 3 November 2015
Basic concept
Basic concept
Basic Sound Reinforcement System
A typical sound reinforcement system consists of; input transducers (e.g., microphones), which convert sound energy into an electric signal, signal processors which alter the signal characteristics (e.g., equalizers, compressors, etc.), amplifiers, which add power to the signal without otherwise changing its content, and output transducers (e.g., loudspeakers), which convert the signal back into sound energy. These primary parts involve varying amounts of individual components to achieve the desired goal of reinforcing and clarifying the sound to the audience, performers, or other individuals.
Some audio engineers
Some audio engineers and others in the professional audio industry disagree over whether these audio systems should be called sound reinforcement (SR) systems or PA systems. Distinguishing between the two terms by technology and capability is common, while others distinguish by intended use (e.g., SR systems are for live event support and PA systems are for reproduction of speech and recorded music in buildings and institutions). In some regions or markets, the distinction between the two terms is important, though the terms are considered interchangeable in many professional circles.
A sound reinforcement system
A sound reinforcement system may be very complex, including hundreds of microphones, complex audio mixing and signal processing systems, tens of thousands of watts of amplifier power, and multiple loudspeaker arrays, all overseen by a team of audio engineers and technicians. On the other hand, a sound reinforcement system can be as simple as a small public address (PA) system, consisting of a single microphone connected to an amplified loudspeaker. In both cases, these systems reinforce sound to make it louder or distribute it to a wider audience.
A sound reinforcement system
A sound reinforcement system is the combination of microphones, signal processors, amplifiers, and loudspeakers that makes live or pre-recorded sounds louder and may also distribute those sounds to a larger or more distant audience.In some situations, a sound reinforcement system is also used to enhance the sound of the sources on the stage, as opposed to simply amplifying the sources unaltered.
Wednesday, 23 September 2015
Applications Tip:
Compressor vs. Loudspeaker
Here is a common complaint made by the owner of a
damaged loudspeaker: "How could I have blown a
loudspeaker? I have a compressor!" Unfortunately, while
compressors and limiters help prevent audio transients
from causing clipping or possibly damaging a loudspeaker,
high-level transients are not the only cause of damaged
loudspeakers. In fact, over-compression of the audio signal
can contribute to premature loudspeaker failure.
It is standard practice to use an amplifier with a power
rating at least twice the loudspeaker’s continuous power
rating (e.g. use a 200 watt amplifier for a 100 watt
loudspeaker). The extra headroom afforded by the larger
amplifier allows for peaks in the program material to be
delivered to the loudspeaker without clipping. The majority
of the amplifier power goes largely unused since the
average level of an uncompressed audio signal is
considerably lower than the peaks. Highly compressed
signals have an average level much closer to the peak
level. If the level of the compressed signal is raised to take
advantage of the additional amplifier power (thereby
making it louder for the audience), the average power
delivered to the loudspeaker may be more than the
continuous power rating of the loudspeaker, overheating
the loudspeaker’s voice coil and causing failure.
As with all audio processors, using a compressor
does not eliminate the need for proper system operation.
Though a compressor or limiter is essential for reducing
transient peaks, excessive compression is the enemy of
the loudspeaker.
Here is a common complaint made by the owner of a
damaged loudspeaker: "How could I have blown a
loudspeaker? I have a compressor!" Unfortunately, while
compressors and limiters help prevent audio transients
from causing clipping or possibly damaging a loudspeaker,
high-level transients are not the only cause of damaged
loudspeakers. In fact, over-compression of the audio signal
can contribute to premature loudspeaker failure.
It is standard practice to use an amplifier with a power
rating at least twice the loudspeaker’s continuous power
rating (e.g. use a 200 watt amplifier for a 100 watt
loudspeaker). The extra headroom afforded by the larger
amplifier allows for peaks in the program material to be
delivered to the loudspeaker without clipping. The majority
of the amplifier power goes largely unused since the
average level of an uncompressed audio signal is
considerably lower than the peaks. Highly compressed
signals have an average level much closer to the peak
level. If the level of the compressed signal is raised to take
advantage of the additional amplifier power (thereby
making it louder for the audience), the average power
delivered to the loudspeaker may be more than the
continuous power rating of the loudspeaker, overheating
the loudspeaker’s voice coil and causing failure.
As with all audio processors, using a compressor
does not eliminate the need for proper system operation.
Though a compressor or limiter is essential for reducing
transient peaks, excessive compression is the enemy of
the loudspeaker.
Compressors
Perhaps the most commonly encountered dynamics
processor, a compressor reduces (or "compresses") the
dynamic range of an audio signal. A compressor functions by
reducing the level of all signals above a user-defined point (the
threshold), by a specified amount. A ratio
defines the amount of reduction that occurs above the
threshold. A ratio of 2:1, for example, will allow an audio
signal to exceed the threshold by only half as much as what it
would have without compression. Assuming a threshold setting
of 0 dB, a +10 dB signal is output at +5 dB. Similarly, a
4:1 setting will reduce the output by one-quarter of the original
signal level. This reduction limits variation between the lowest
and highest signal levels, resulting in a smaller dynamic range.
A common myth concerning compressors is that they make
quiet signals louder. While this may be the perceived effect,
reducing the dynamic range of a signal allows the user to
boost the overall level of the signal, yet keeps loud signals from
getting "too loud" and causing distortion further down the
audio chain - or simply annoying listeners. The compressor
itself does not boost lower signal levels, but simply allows them
to be perceived closer in level to louder signals.
Other compressor settings include attack, release, and
decays. A compressor’s attack time relates to how quickly the compression takes effect once the signal exceeds thethreshold. Shorter attack times offer greater transient control.
Longer attack times generally sound more natural, and are
often employed in musical applications. Too long an attack
time can cause the compressor to miss signals that otherwise
should be compressed. Release refers to the time it takes for
the compressor to return the signal level to its original value
after the level drops below the threshold. Too short a release
time can result in "pumping" and "breathing" with signals that
have rapid level changes. Too long a release time can render
quieter passages inaudible since gain reduction is still being
applied to the audio signal.
A compressor’s knee is the point where the signal
crosses the threshold. Standard compression schemes
reduce the signal by the same amount once the signal has
passed the threshold. This is known as hard knee
compression. Some compressors allow the user to select
soft knee compression instead, where the onset of
compression near the threshold occurs more gradually than
the more aggressive hard knee compression. The compression ratio near the threshold is actually
less than specified. Audibly, soft knee compression creates
a more gradual transition from uncompressed to
compressed signals, making the compression less noticeable.
processor, a compressor reduces (or "compresses") the
dynamic range of an audio signal. A compressor functions by
reducing the level of all signals above a user-defined point (the
threshold), by a specified amount. A ratio
defines the amount of reduction that occurs above the
threshold. A ratio of 2:1, for example, will allow an audio
signal to exceed the threshold by only half as much as what it
would have without compression. Assuming a threshold setting
of 0 dB, a +10 dB signal is output at +5 dB. Similarly, a
4:1 setting will reduce the output by one-quarter of the original
signal level. This reduction limits variation between the lowest
and highest signal levels, resulting in a smaller dynamic range.
A common myth concerning compressors is that they make
quiet signals louder. While this may be the perceived effect,
reducing the dynamic range of a signal allows the user to
boost the overall level of the signal, yet keeps loud signals from
getting "too loud" and causing distortion further down the
audio chain - or simply annoying listeners. The compressor
itself does not boost lower signal levels, but simply allows them
to be perceived closer in level to louder signals.
Other compressor settings include attack, release, and
decays. A compressor’s attack time relates to how quickly the compression takes effect once the signal exceeds thethreshold. Shorter attack times offer greater transient control.
Longer attack times generally sound more natural, and are
often employed in musical applications. Too long an attack
time can cause the compressor to miss signals that otherwise
should be compressed. Release refers to the time it takes for
the compressor to return the signal level to its original value
after the level drops below the threshold. Too short a release
time can result in "pumping" and "breathing" with signals that
have rapid level changes. Too long a release time can render
quieter passages inaudible since gain reduction is still being
applied to the audio signal.
A compressor’s knee is the point where the signal
crosses the threshold. Standard compression schemes
reduce the signal by the same amount once the signal has
passed the threshold. This is known as hard knee
compression. Some compressors allow the user to select
soft knee compression instead, where the onset of
compression near the threshold occurs more gradually than
the more aggressive hard knee compression. The compression ratio near the threshold is actually
less than specified. Audibly, soft knee compression creates
a more gradual transition from uncompressed to
compressed signals, making the compression less noticeable.
Types of Audio Processors
crossover feeding three power amplifiers is called a triamplified
system. If clipping occurs in the low frequency
amplifier, the higher frequency harmonics created by the
clipping are reproduced only by a woofer that has very low
output at high frequencies, thus reducing the
audibility of the distortion. The use of active components
also offers smaller size and more repeatable production
due to better tolerances.
Quite often, a sound system combines elements of
both passive and active crossover networks. These types
of systems typically use an active crossover to provide a
separate subwoofer output for low frequencies, while a
passive crossover in a two-way loudspeaker divides
mid- and high frequencies. This could be described as a
three-way, bi-amplified sound system.
Most active crossovers allow for control of crossover
frequency and level at each output. DSP-based crossovers
typically offer greater adjustment, providing the user with
selectable filter slope, filter type, and polarity reversal.
system. If clipping occurs in the low frequency
amplifier, the higher frequency harmonics created by the
clipping are reproduced only by a woofer that has very low
output at high frequencies, thus reducing the
audibility of the distortion. The use of active components
also offers smaller size and more repeatable production
due to better tolerances.
Quite often, a sound system combines elements of
both passive and active crossover networks. These types
of systems typically use an active crossover to provide a
separate subwoofer output for low frequencies, while a
passive crossover in a two-way loudspeaker divides
mid- and high frequencies. This could be described as a
three-way, bi-amplified sound system.
Most active crossovers allow for control of crossover
frequency and level at each output. DSP-based crossovers
typically offer greater adjustment, providing the user with
selectable filter slope, filter type, and polarity reversal.
FILTERS AND EQUALIZATION
Filters are signal processors that affect frequency
balance. At a basic level, filters are used to attenuate or
boost the level of specific frequencies or frequency ranges.
Designed originally to compensate for frequencydependent
loss in telephone lines, some form of frequencydependent
filtering (or equalization) is found in all but the
most basic of sound systems. The simplest form of filter is
the tone control, basically a filter that attenuates high frequencies
above a predetermined frequency. Equalizers are
typically characterized by combining several filter sets to
offer more precise frequency response shaping.
Historically, filters were passive devices capable of
attenuation only. The frequency range and amount of
attenuation were achieved with capacitors, inductors, or a
combination of both. Favorably, passive filters do not
require power and do not generate noise. The large size
and expense of discrete components, however, precludes
the ability to develop equalizers with larger numbers of
filters and more precise control of frequency and level.
Active filters allow for fast, easy tuning and the ability to
add gain, using smaller components at lower cost. Tone
controls employing active filters can be found on even the
most inexpensive home stereo systems. In this scenario
there are typically two controls, treble and bass, which
correspond to filters that affect low frequency and high
frequency response. Since they are active, these tone
controls are capable of cut or boost.
Simple filters that affect a broad range of frequencies
are divided into four types: high pass, low pass, band
pass, and band reject. High pass filters, as the name
implies, allow high frequencies to pass, and low pass filters
do the same for low frequencies. It is often more convenient
to think of these filters in terms of the frequencies that they
cut instead. High pass filters are also known as low cut
filters, and low pass filters are known as high cut filters, but
their function is the same and these terms can be used
interchangeably. Low and high cut filters
have an associated slope that defines how rapidly output
declines below (or above) the filter frequency. Slope is
typically defined in dB/octave. The span of an octave
relates to a doubling (or halving) of frequency, for example,
50 to 100 Hz or 5 kHz to 2.5 kHz. A 6 dB/octave low cut
beginning at 100 Hz, therefore, translates into 6 dB less
balance. At a basic level, filters are used to attenuate or
boost the level of specific frequencies or frequency ranges.
Designed originally to compensate for frequencydependent
loss in telephone lines, some form of frequencydependent
filtering (or equalization) is found in all but the
most basic of sound systems. The simplest form of filter is
the tone control, basically a filter that attenuates high frequencies
above a predetermined frequency. Equalizers are
typically characterized by combining several filter sets to
offer more precise frequency response shaping.
Historically, filters were passive devices capable of
attenuation only. The frequency range and amount of
attenuation were achieved with capacitors, inductors, or a
combination of both. Favorably, passive filters do not
require power and do not generate noise. The large size
and expense of discrete components, however, precludes
the ability to develop equalizers with larger numbers of
filters and more precise control of frequency and level.
Active filters allow for fast, easy tuning and the ability to
add gain, using smaller components at lower cost. Tone
controls employing active filters can be found on even the
most inexpensive home stereo systems. In this scenario
there are typically two controls, treble and bass, which
correspond to filters that affect low frequency and high
frequency response. Since they are active, these tone
controls are capable of cut or boost.
Simple filters that affect a broad range of frequencies
are divided into four types: high pass, low pass, band
pass, and band reject. High pass filters, as the name
implies, allow high frequencies to pass, and low pass filters
do the same for low frequencies. It is often more convenient
to think of these filters in terms of the frequencies that they
cut instead. High pass filters are also known as low cut
filters, and low pass filters are known as high cut filters, but
their function is the same and these terms can be used
interchangeably. Low and high cut filters
have an associated slope that defines how rapidly output
declines below (or above) the filter frequency. Slope is
typically defined in dB/octave. The span of an octave
relates to a doubling (or halving) of frequency, for example,
50 to 100 Hz or 5 kHz to 2.5 kHz. A 6 dB/octave low cut
beginning at 100 Hz, therefore, translates into 6 dB less
VOLUME (GAIN) CONTROL
Although often overlooked as an audio processor, a
simple volume (or gain) control fits the definition. Volume
adjustments can be made at several points within the
sound system, from the microphone inputs on the mixer all
the way to the inputs of the power amplifiers. Volume
levels are typically manipulated in one of two ways:
continuously variable adjustments, such as those made by
rotary potentiometers or faders, or fixed attenuation such
as that provided by a pad.
If adjusting a volume control adds amplification to the
audio signal, it is said to be providing gain. The volume
control that adjusts the amount of amplification added at a
mixer’s microphone input is sometimes referred to as a
gain (or trim) control, since the volume potentiometer is
controlling the gain of the microphone input’s preamplifier.
The function of this gain control is to match the input
sensitivity of the device to the level from the source.
A second type of volume control acts as an
attenuator, basically a continuously variable resistor that
adjusts the amount of signal allowed to pass through it. No
additional gain is provided by the volume control. The
volume control on an electric guitar is an attenuator. These
devices are often referred to as passive volume controls,
since they do not require any power. Occasionally, a
volume control will combine attenuation with gain. Faders
on a mixing console typically provide attenuation below
the "0" indication, and gain above that point.
Pads allow input stages to accommodate a variety of
signal levels. Microphone inputs typically feature an input
attenuation pad of some kind to reduce the sensitivity of
the input beyond that of the preamplifier gain control,
typically by 20 dB. A 50 dB pad is required for microphone
inputs that are designed to accept either microphone or
line level. The output stage of various devices can also
employ pads, usually to prevent overloading of the input
stage of the next device in the signal path. Care should be
taken to use pads only when necessary. For example,
using a 20 dB pad on a microphone input that does not
need additional attenuation will require additional gain be
added by the preamplifier, which adds more noise to the
audio signal.
While volume controls are the simplest of all audio
processors, they often the most misused. Correct
calibration of the various volume controls in a sound
system is known as proper gain structure.
simple volume (or gain) control fits the definition. Volume
adjustments can be made at several points within the
sound system, from the microphone inputs on the mixer all
the way to the inputs of the power amplifiers. Volume
levels are typically manipulated in one of two ways:
continuously variable adjustments, such as those made by
rotary potentiometers or faders, or fixed attenuation such
as that provided by a pad.
If adjusting a volume control adds amplification to the
audio signal, it is said to be providing gain. The volume
control that adjusts the amount of amplification added at a
mixer’s microphone input is sometimes referred to as a
gain (or trim) control, since the volume potentiometer is
controlling the gain of the microphone input’s preamplifier.
The function of this gain control is to match the input
sensitivity of the device to the level from the source.
A second type of volume control acts as an
attenuator, basically a continuously variable resistor that
adjusts the amount of signal allowed to pass through it. No
additional gain is provided by the volume control. The
volume control on an electric guitar is an attenuator. These
devices are often referred to as passive volume controls,
since they do not require any power. Occasionally, a
volume control will combine attenuation with gain. Faders
on a mixing console typically provide attenuation below
the "0" indication, and gain above that point.
Pads allow input stages to accommodate a variety of
signal levels. Microphone inputs typically feature an input
attenuation pad of some kind to reduce the sensitivity of
the input beyond that of the preamplifier gain control,
typically by 20 dB. A 50 dB pad is required for microphone
inputs that are designed to accept either microphone or
line level. The output stage of various devices can also
employ pads, usually to prevent overloading of the input
stage of the next device in the signal path. Care should be
taken to use pads only when necessary. For example,
using a 20 dB pad on a microphone input that does not
need additional attenuation will require additional gain be
added by the preamplifier, which adds more noise to the
audio signal.
While volume controls are the simplest of all audio
processors, they often the most misused. Correct
calibration of the various volume controls in a sound
system is known as proper gain structure.
Saturday, 1 August 2015
Figure of Eight
This pattern is used mainly in studios for picking up two ‘harmony’
vocalists, or solo vocalists who require some room ambience.
Hyper-cardioid
with greater directionality.
Used for live vocal microphones
because it provides the greatest
protection from unwanted spill
and feedback.
Cardioid Pattern
recording or in any situation where sound has to be picked up from mainly one direction. Dynamic cardioid mics are mostly used for live
applications because they help reduce unwanted spill
from other instruments, thus reducing the risk of
feedback.
Omni Pattern
microphone pattern.
• A 360° polar response which
picks up sound equally in all
directions.
This pattern is ideal for
picking up groups of vocals,
audiences, ambient sounds but is most susceptible to feedback.
Microphone Pick-up Patterns
microphone "picks up" its sound. It is important to choose
the right pattern for your application, or you may pick up
sounds from areas you don’t want or lose sound
information you need.
Tuesday, 28 July 2015
Applications Tip: Compressor vs. Loudspeaker
Here is a common complaint made by the owner of a
damaged loudspeaker: "How could I have blown a
loudspeaker? I have a compressor!" Unfortunately, while
compressors and limiters help prevent audio transients
from causing clipping or possibly damaging a loudspeaker,
high-level transients are not the only cause of damaged
loudspeakers. In fact, over-compression of the audio signal
can contribute to premature loudspeaker failure.
It is standard practice to use an amplifier with a power
rating at least twice the loudspeaker’s continuous power
rating (e.g. use a 200 watt amplifier for a 100 watt
loudspeaker). The extra headroom afforded by the larger
amplifier allows for peaks in the program material to be
delivered to the loudspeaker without clipping. The majority
of the amplifier power goes largely unused since the
average level of an uncompressed audio signal is
considerably lower than the peaks. Highly compressed
signals have an average level much closer to the peak
level. If the level of the compressed signal is raised to take
advantage of the additional amplifier power (thereby
making it louder for the audience), the average power
delivered to the loudspeaker may be more than the
continuous power rating of the loudspeaker, overheating
the loudspeaker’s voice coil and causing failure.
As with all audio processors, using a compressor
does not eliminate the need for proper system operation.
Though a compressor or limiter is essential for reducing
transient peaks, excessive compression is the enemy of
the loudspeaker.
Limiters
compressor, differentiated more by its application than its
operation. Similar to a compressor, a limiter also reduces
signals that pass a threshold by a certain ratio. The ratios used
by limiters, though, tend to be much greater than those used
by compressors. Typical limiter ratios can range anywhere
from 10:1 to ∞:1 (infinity:1, where the threshold setting dictates
the maximum signal level). The goal of
a limiter is usually system protection, by preventing transient
audio peaks from causing distortion further up the audio chain
or, worst case, damaging loudspeaker components. Typically,
limiter threshold settings are also much higher than on
compressors; low threshold settings on a limiter lead to excess
compression. Limiters also share other parameters with
compressors, including attack and release.
Graphic Equalizers
reinforcement is the graphic equalizer. A typical graphic
equalizer consists of a bank of sliders (or faders),
corresponding to specific frequencies, which can cut or
boost the chosen frequency. The center
frequencies of these filters are identical for all graphic
Sunday, 26 July 2015
Poor Speech Intelligibility
Bad acoustics and/or an inadequate sound system can
make speech difficult to understand. Everyone can hear
the preacher, but most can't understand, or at best have
to strain to understand. The result is wandering attention
to the message and poor communication between
the praise and worship team and the congregation.
Worshippers feel isolated from the mass; their understanding
of the message, lessons, and announcements
is hampered and their participation is more distant.
This can directly impact attendance and contributions
to every phase of parish life
Acoustics
Bad acoustics can make music sound bad and require
a more expensive sound system. When this is the case,
music will never sound good until the acoustics are
fixed, often at great cost and with a big mess during
construction.
The Acoustics
consultant and the architect work together.
• It is always cheaper and better to do it right the first
time.
• If budgets are limited, it is far better to have the design
work done correctly and defer the purchase of
options such as an organ or a sound system.
• Acoustic and sound system consultants are good
insurance against serious problems later on.
• Acoustics and sound system design are applied
physics, not “black magic.” Poor acoustics and sound
system performance means either that the design
team made mistakes or their recommendations were
not properly implemented.
• Good acoustics rarely happen by accident. Poor
acoustics are a far more likely result when there is no
consultant on a project.
• No sound system can remove reverberation from a
space. If contemporary music will be part of the worship,
reverberation must be carefully controlled.
• Contemporary music and traditional European music
make very different and conflicting demands on
room acoustics. It is quite costly (and often impractical)
to provide a workable acoustic environment for
both forms in the same space.
• Any architect who knows anything about acoustics
will hire a good acoustic consultant for any church
or other large space. The opposite is also true — any
architect who doesn't hire a good acoustic consultant
for these spaces thereby demonstrates his
incompetence.
• Some of the worst acoustic environments I've ever
experienced were designed on the advice of an organ
builder. Not only were they nearly impossible for the
spoken word, they were awful for for the choir, congregational
singing, and other musical components
of worship.
Acoustic consultants
Acoustic consultants work with churches and their
architects to provide an acoustic environment that fits
their form of worship. When a church is being built
from the ground up, this means helping determine
the size and shape of the worship space, as well as the
choice and placement of finishes and furnishing within.
It also means looking over the design of mechanical
systems and room layouts to insure that the church is
quiet and free of interfering noise. If the church building
already exists, it may mean major rebuilding and/or
refinishing of the worship space.
architects to provide an acoustic environment that fits
their form of worship. When a church is being built
from the ground up, this means helping determine
the size and shape of the worship space, as well as the
choice and placement of finishes and furnishing within.
It also means looking over the design of mechanical
systems and room layouts to insure that the church is
quiet and free of interfering noise. If the church building
already exists, it may mean major rebuilding and/or
refinishing of the worship space.
Getting the Acoustics and Sound System Right
The factors that make for good sound in a church need
to be built into the design from the beginning. Since
the acoustics of a space are so highly dependent on its
shape and finishes, improving the acoustics after the
building is completed is often a very expensive proposition.
Having the proper loudspeaker and microphone
systems installed in the right place is fundamental to
good sound system performance. If the architect has not
provided for the right loudspeaker system in the original
building design, the congregation has to make a
choice between looking at ugly loudspeakers that work
well, not seeing loudspeakers that don't, or spending a
lot of money to hide them after the fact.
Not all churches need the same kind of acoustic
environment. Good acoustics for one congregation
may be unsatisfactory for another. A Gothic structure
is a wonderful environment for a congregation whose
normal liturgy is rooted in European music and Gregorian
chants, if it has a good sound system to make
speech intelligible. The same church would be unusable
for contemporary Christian music, and wouldn't work
very well for a jazz mass or a choir in the contemporary
gospel tradition.
Sound Systems as Problem Solvers
Well-designed sound systems can help overcome
acoustic problems and enhance the worship experience.
To work well, they must work hand-in-hand with
room acoustics. They are used in churches in three
basic ways.
1. Sound systems make sound louder, so that a weak
voice or musical instrument can fill the church without
great effort. They can reach into distance seating
areas such as those in and under balconies where
worshippers would otherwise feel isolated or have
difficulty hearing.
2. Sound systems can provide speech intelligibility in
spaces that would otherwise be too reverberant. The
right kind of sound systems can eliminate the acoustic
conflict between music and speech by bringing
amplified speech more directly to the listener
without allowing it to bounce around the walls of the
worship space in an uncontrolled way. The church is
still reverberant; its acoustics can support traditional
church music that demand reverberation. The sound
system provides an additional means of controlling
the sound. Music which needs reverberation does
not utilize the sound system, taking advantage of the
acoustics of the room.
3. Sound systems can sometimes allow contemporary
music to be effective in an environment than would
otherwise be too reverberant. They do this in the
same way that they control speech — by controlling
the amplified sound, carefully focusing it on listeners
rather than allowing it to be turned into reverberation
by the acoustics of the space. Sound systems
that will be used for contemporary music need to
be capable of providing the naturalness, impact and
dynamics the form requires.
Tuesday, 7 July 2015
LCR Panning – an introduction to three-channel live systems
There is stereo panning and there is LCR panning. So what’s the difference and why would you want to
choose LCR panning? After all, it adds quite a cost to a console and stereo is stereo right?
Well, not really. Let’s just remind ourselves that Stereo panning simply adjusts the amount of signal
sent to the left and right outputs. The actual mix output may not be totally stereo, and probably
includes a high degree of spoken words as well as singing which is mono. This is particularly true in
Dramatic performances and Houses of Worship, and being able to pan voices to their true stage
position can be difficult in straight stereo. This is because we are always balancing the voices between
the left and right speaker positions. Some voice will always come out of both speakers unless panned
hard one way or the other.
Saturday, 4 July 2015
Although monitor engineering is often thought of as subordinate to handling the FOH sound, in reality it's at least as important.
To
create a musical performance, two things have to happen: your audience needs to
hear you, and you need to hear yourself. If you can't hear yourself clearly,
how will you know that you're playing or singing well? In purely acoustic
music, being able to hear oneself is often taken for granted. But there are
situations where this doesn't happen as it should. For example, in an orchestra
performing on stage, each musician needs to hear his or her own instrument
clearly and distinctly from the other instruments around them. But sometimes
the acoustics on stage make this difficult. Suddenly, one's ability to perform well
has been diminished severely by the inability to hear one's own playing.
The
same applies to amplified music. In the early days of pop and rock bands it was
common to provide only front-of-house (FOH) amplification, commonly mixed from
the stage by a member of the band (who, of course, couldn't hear the FOH PA
properly). Although many exciting performances (and undoubtedly many distinctly
unexciting ones) have been given in this way, the fact is that no-one is
properly in control of what the audience hears. The one advantage is that the
band can angle the speakers and set their levels so that they can hear
themselves and each other reasonably well, most of the time.
Fortunately,
progress has been made and we now recognise that it is essential to have the mix
position at front-of-house, placed centrally amidst the audience area. The FOH
engineer is now ideally placed to control the sound the audience hears. The
problem now is that the band are no longer in any kind of control whatsoever of
what they hear. Clearly, in an ideal scenario, there should be a completely
independent system to provide the band with crystal clear sound so they can
hear their own individual performances and the overall sound of the ensemble.
This is what stage monitoring should provide.
Stage
monitoring is taken very seriously by top professionals, and should be by
anyone working in live performance, right down to pub gig or theatre foyer
level. Good monitoring consists of having the right equipment, suitable for the
nature of the venue and performance, setting it up well and, of course,
operating it effectively.
Above
all, musicians need to feel that they are making great sounds. If they feel
that the performance is good, the performance will be good and the audience
will go away whistling the tunes. Also, performers need to feel secure.
Security comes from knowing what the other band members are doing, knowing
where they are in the song, and being certain that the notes and rhythms they
are playing fit in with the rest of the band.
So
let's imagine you're the lead singer of a band. The lead singer needs to feel
that his or her voice is strong, in tune, and communicating emotion to the audience.
Clarity and good tone of voice are paramount. Also, the lead singer needs to
hear the band, so that they know they are in tune and are fully comfortable
that the band are following them precisely. If the band are playing to a click
track or a recorded backing, strict tempo will be an issue and the lead singer
may need the band to be more emphasised in the foldback, since now everyone has
to follow the click (even though only the drummer would normally hear it) or
recording; the band cannot follow the singer.
The
other band members have their own individual requirements, but in general they
also need to feel that they sound great. They need to hear the vocal, too,
otherwise they might have a blank moment and forget whether they're in verse
two or verse three (that's scary when it happens). They will also have a
preference about which other instruments they need to hear most clearly, to
feel as though they're 'gelling' with the rest of the band.
Friday, 3 July 2015
20 Tips On Mixing
resting your ears overnight. Also check the master recording on as many different sound systems as you can,
to ensure it sounds fine on all of them. Even then, save all your mix information and track sheets, including
effects settings, as you never know when you might want to try to improve on the 'final mix'!
20 Tips On Mixing
sound seems unnaturally lifeless, but you don't want to add any obvious reverb, try an
ambience or early reflection setting to induce a sense of space. The shorter the reverb time, the easier it is to
move the treated sound to the front of your mix.
20 Tips On Mixing
track, try not to look at your sequencer display while mixing; the
visual stimulus interferes with your ability to make subjective judgements based only on the sound. If
necessary, close your eyes. Watching your sequencer progress through the arrange page can also give you a
false impression of how well the arrangement is working, which is why some composers prefer hardware
sequencers.
20 Tips On Mixing
instruments such as overdrive guitars and synth pads under the control
of the vocals, so that whenever the vocals are present, the conflicting sounds fall in level by two or three dBs.
Just a little ducking can significantly improve the clarity of a mix. Use a fairly fast attack time for the ducker
(which may be either a compressor or a noise gate that has ducking facilities), and set the release time by ear.
Shorter release times will cause more obvious gainpumping,
but in rock mixes, this can add welcome energy
and excitement
20 Tips On Mixing
the constant backdrop against which other sounds move. Natural dynamics within rhythm instrument parts is
OK, but don't keep moving the faders on these sounds.
20 Tips On Mixing
you may never hear over loudspeakers. However, don't rely solely on headphones for mixing, for they
represent the stereo image differently to loudspeakers and are notoriously unpredictable at low frequencies.
20 Tips On Mixing
listen at the same high level. High monitoring levels also tend temporarily to shift your hearing perspective and
can lead to permanent hearing damage. It's fine to check the mix loudly for short periods, but most of the time,
it's useful to try and mix at the level you think the music will eventually be played. (Forget I said this if you're
mixing dance music for nightclubs!)
20 Tips On Mixing
show up level imbalances more clearly than when listening from directly in front of the monitors. Nobody is
quite sure why, but it works.
20 Tips On Mixing
be mixed successfully without compression. Softknee
compressors tend to be the least obtrusive, but if you
want the compression to add warmth and excitement to your sound, try an optocompressor
or a hardknee
model with a higher ratio setting than you'd normally use. Be aware that compression raises the background
noise (for every 1dB of gain reduction, the background noise in quiet passages will come up by 1dB), and
heavy compression can also exaggerate vocal sibilance.
20 Tips On Mixing
cut than it is to boost. This is especially true if you are using a lowcost
equaliser or the EQ in your desk.
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