Lenard Education

Mixing and Production

The Subjective Craft  of mixing is mastered by knowing when to leave the knobs alone.   More blood has been spilt by this oversight than all other problems combined.   The ability to be excellent in the craft of production arrangement (mixing) is a gift similar to the craft of musical arrangement.

There are hundreds of web sites, books and periodicals with excellent information on mixing and production.   This page is not intended to repeat information, but to prioritise the order of information (especially hardware understanding) essential for best practice.   This page also challenges many of the inept practices and false beliefs entrenched within the pop recording industry.

The above mixer design evolved from the film and broadcast industry.   Note the small monitor speakers on top of the mixing desk.   These mixers can be used by many operators as a production team.   Each operator managing a different section of the sound track.   Editing sync,  dialogue,  background environments,  effects and music.

These large production consoles became used by the pop recording industry from the late 1960s.   A large percentage of the pop music industry is image driven and recording studios competed by who had the LARGEST mixing console and multi-track machine.   By the 80s almost every studio had variations of large consoles.   Competition then became driven by the latest models of compressors EQs and effect units.

Many pop recording budgets were and are small.   Approx 1,000 hours of studio time may result in less than 1 hour of recorded music heard by the public.   Less than 1 in 4 pop recording studios survive the first year, producing a large bankrupt stock of second hand consoles at low cost.   The enemy is ego, junk food, caffeine,  religious attachment to brand names, and loss of sight of the fundamentals.

The above pic represents the new automated digital consoles, ergonomically designed for a single operator.   A skilled operator can achieve the same outcome in a fraction of the time as the larger designs in the previous pic.   These smaller designs evolved in the 1980s but were not accepted by pop recording studios because of their small image.

Pop Recording

The traditional art of pop recording is to   'Construct the balance of a performance'.   Reducing the dynamic range of the music, and compacting it with added effects, that enable it to sound acceptable on small, low fidelity domestic sound systems.   The pic below is of a recording studio with small monitor speakers similar to many domestic sound systems.

Recording today is as it was in the 1950s 1960s, compacting (compressing) the music to be heard best on small limited fidelity 4" to 8" speakers.   Also constrained within the limited bandwidth of radio, TV, internet, and MP3.   This requires the skilled use of effect processors.

• Frequency based    -   equalization, filters etc.
• Time based              -   delay, reverberation etc.
• Amplitude based     -   compression limiting, noise gates etc.

It is not possible for small, low fidelity domestic sound systems to replicate the full dynamic power of a real orchestra.   Live music can have a dynamic power greater than 60dB (1 : 1,000,000).

The average domestic sound system may handle 20 Watts before it distorts.   Approx 100 mW (1/10 Watt) is required to hear music at low level, depending on background ambient noise.   The ratio of power difference between 1/10 Watt and 20 Watts is 1 : 200 (23 dB).   The majority of recordings are compressed within 20 dB and many pop recordings compressed within 10 dB.

Phil Spector   Wall of sound

Music of the 1950s and 60s   was predominantly recorded to be heard on small portable record players and radiograms.   The same size as domestic systems of today.   With modern computers the recording technology has greatly improved but what it is used for has not changed.   Today's small sound systems may have a slightly broader frequency response and more power but the overall auditory experience is very similar.   The beautifully restored Blaupunkt radiogram below could be argued as having superior fidelity than many of todays small sound systems.

Phil Spector and the wall of sound   is the name and term associated with pioneering production techniques (50s - 60s) giving a perception of a larger sound being experienced with small speakers.   These production techniques also enabled maximum use of the limited dynamic range and bandwidth of AM radio transmission.

Google search on   (Phil Spector)   (Mowtown)   (Recording history)

There are many examples of recorded music, live bands, TV and film sound   that has been excellently produced with the correct use of compression and EQ.   However little is said of pop mixing practices, especially the over use of compression that is destructive to the enjoyment of music.   Please accept this criticism of the miss-use of excessive compression on behalf of all who are passionate of best practice.

The pop recording industry is dominated by  trends,  fads  and brand names  often with limited understanding of hardware technology,   it is held back by 3 fundamental beliefs.

(a)   Classical training or learning to read music will destroy my feel for playing pop.
(b)   Basic technical knowledge of 'dB' and hardware is not required for mixing pop.
(c)   The good sound knobs are   Compression   Compression   Compression

Many pop recording schools make money from young people supporting these beliefs.   Playing with audio mixing computer packages and repeating jargon and brand names has become an education in itself   including that 'compression' solves all problems and makes everything sound wonderful.

The Past   Critical statements are often made about record companies failure to understand the digital revolution and the internet.   Management practises based on short term thinking often have difficulty seeing opportunity's to transform their industries when new technology arrives.   We must remember that record companies also strongly resisted the introduction to stereo recording during the 1950s.

Future opportunity   The traditional over-compressed music for small speakers should be freely available over the internet.   The irony is that by piracy it already is,   reflecting its true value.   Full dynamic minimally-compressed music arranged by different producers , as mixed stereo, or as original multi-tracks, should be available at an agreeable price.   The opportunities created could be endless.   With full fidelity active sound systems and advanced software for production mixing available to home users, the number of people participating could be magnitudes greater than it is today.   The economic turnover would be an extension and similar to the computer games industry.

Live mixing

Young musicians and bands identify with iconic brand names (Fender Marshall etc) representing 1950s technology from the beginning of rock and roll, and very little has progressed since that time.   It is essential for live music to progress past this 1950s approach and take advantage of modern technological management if it is to economically thrive.

The above pic is of a band comparing many different PA's.   This traditional mixing approach is the same for domestic recording.   The PA speakers are set up to represent a domestic sound system on a larger scale.  Imposing the techniques of recording on a live production by constructing a separate compacted (compressed) mix to be heard on a poor fidelity and often distorted PAs, leads to the sound system competing with the performance.   The result is often a cluttered mess which has little resemblance to how live musicians naturally sound up close without the PA.

This miss-use of technology which separates the audience from the performance has possibly contributed more to the downfall of live performances than all other problems combined.

The art of live mixing   should be the opposite to traditional mixing for pop recording,   e x p a n d i n g   the expression of the music,  bringing the performance forward into the audience.   However, the musicians and artists must be in balance first.   Bringing the performance forward with minimal change to the musical balance and with minimal use of effects,   especially equalization and compression.   This requires a greater discipline and understanding of

• Sound systems    -   speakers, amplifiers etc.
• Acoustics             -   room reverberation, critical-distance, directivity etc.
• Microphones       -   types, placement, direct and transmitting etc.

The first commandment   of live mixing "Thou cannot equalise time with amplitude".   As punishment thou shall be plagued with microphone feedback problems.   Trying to compensate for bad acoustics and poor fidelity speaker systems with equalization is as silly as a dog chasing its tail.

The above pic of   The Who 1969   shows instrument amps and drums at front of stage.   The audience could hear the band directly.   These performances were often more energetic and louder than live bands today.   This live approach of hearing the band directly only lasted a short time, approx 10 yrs.   It was considered too difficult and complex to increase this method for large concerts, as in the pic below.

Click on the link above to see the 1970's The Grateful Dead sound system in full size on the Altec heritage site.   This above approach had voice, instruments and percussion amplified and heard separately.   This technique was more complicated and ahead of its time.   However the emerging IT industry consumed the best technical people where they could earn 10 - 100 more than working in the entertainment industry.   This is the primary reason this method did not become developed for live music production.

Future opportunity   By the use of visual analogy the first step is to unlock the fixed psychological perception of a compressed mono mixed, often distorted sound.   The grandeur of a large scale firework displays is used in the below pic to demonstrates this.

Watching a   small mono   video does not create the same experience for the observer as seeing the actual fireworks.   When watching a video screen it is only the video screen that is being seen.

This may appear obvious,   but it is strange that many people don't get this.   The mind has to make an abstract interpretation of a video representing fireworks.   The video screen is a flat 2D image at one point in space.   The actual fireworks appear in 3D space from an infinite number of points.

Now we shall repeat the same example with music.   Acoustically we hear a live orchestra as an infinite multitude of sound sources as a dynamic panoramic experience.   When listening to a live performance through a low-fidelity PA system, mixed in mono and compressed, it is only the Left or Right speakers that are heard, not the actual live performance.

The audience only hear the speaker that is closest to them, and hear no center image or stereo depth of field.   This is the same non-exciting experience as watching a video while pretending to imagine the fireworks.   Also this is dual mono and not to be confused with stereo.

Stereo sound   dictionary definition adj:   reproducer in which 2 microphones feed two or more loudspeakers to give a 3D three-dimensional effect to the sound.   Stereo vision is 2 photographs taken from slightly different angles that appear three-dimensional when viewed together.   Stereo is therefore the resultant 3D experience.   Not to be confused with dual mono (left and right).

A multi channel of 4 or 5 full-fidelity sound system can approximate a live orchestra, creating a virtual reality 3D experience for the audience.   The aim is to enable the mixing process to be a transparent as possible.   This means that the microphones appear to be directly connected to the speaker system.   The sound system must be full fidelity with minimum to zero compression, for the 3D effect to be experienced.

Quad stereo   can create 6 stereo fields from 4 speaker systems (AB) (AC) (AD) (BC) (BD) (CD).   With a full fidelity active sound system each pair is able to function in full stereo to create a 3D panoramic depth of field.   This can be increased to any number and is only limited by ones imagination and ability to manage the mixing complexity.

The microphones should appear to be connected directly to the amplifier speaker system, without the mixing process being apparent.   This transparent mixing procedure will be discussed in more detail on the Microphones page.

The same experience can be created for a pop/jazz/rock production to enable the audience to experience the live music similar to being on stage with the musicians.   This also requires a multi or quad sound system, based on full fidelity active technology.   The production for live music should be separating the voices and instruments into an exciting wide panoramic spatial sound scape where the sound creation (not lighting) becomes the primary experience for live music.

There are infinite variations on how this can be achieved.   The variations should be tailored to each production.   This approach has always been applied to lighting.   With the assistance of digital management technology, now is the time for creative sound sculpturing to take center stage.

Cinema Sound

Mixing for cinema requires all previous skills.   The objective is to match the sound with the moving image, enabling the picture to come alive.   This requires sound to carry detail represented in the picture.   If well done, the sound caries the moving image from 2 to 3 Dimensions.   This effect is called Synesthesia.

Synesthesia   is the experience we have when one sensory system effects another.   A passing scent can evoke a strong memory,  we can experience colours as warm-red or cold-blue.   The music score and sound effects influence our feelings and how we experience seeing the picture.   Synesthesia stems from the function of our imagination carrying sensory experience into altered states.   Capturing one of our senses (sound) into 3D experience caries the other senses with it, giving the picture a larger 3D perception and enhance our enjoyment of the movie.

Approx 1 in 30 films are created by people who have this understanding and skill.   The dialogue leads with visual form edited to the music score,  'Form following content'.   This procedure is fully thought out in pre-production.   Often the (left-center-right) speakers are utilised as a tri stereo field giving a full auditory depth of field to the picture.

Unfortunately the majority of   small 5.1 home cinema sound systems   are of poor fidelity, causing the synesthesia effect to be limited or not possible.   A large full-fidelity active stereo system will sound more realistic and swamp a small 5.1 system rendering it effectual.   Also the small low fidelity domestic 5.1 systems have little in common with large commercial cinema sound systems.

The Academy Characteristic   is the standard protocol from early cinema history established in the late 1930s which included EQ per-emphasis to compensate for noise on film stock and the limited fidelity of cinema speakers.   The standard also required the center speaker to carry the majority of the story information, without the need for left right and surrounds.   Trapped by its history, cinema sound is fundamentally a mono format, which later added left right and surrounds.   This regulation is not mandatory and mixing sound for cinema has always been add-hock.

http://www.filmsound.org   is one of many excellent sites with historical information on cinema sound.

Many cinema complexes struggle to exist.   A yearly average occupancy rate can be below 15%.   Understandably many cinema complexes have little economic interest in supporting sound quality or surround sound formats,  as they believe this has no influence on audience attendance.   Through cost cutting one technician may have to look after many cinema complexes in a city region.   The different audio formats which need to be checked for each film may be defaulted to the simplest system, or deferred to mono.

With all the good intentions the majority of films for the 12 to 25 yr old junk market are for short term cinema viewing before going to the video market.   Many of these films are constructed with dialogue and music in a post production hotch-potched manner.   The result is often a quasi mono compressed sound with disjointed periodic surround effects.

Dumbing down   Regardless of how well a script is written, excessive dynamic compression makes the dialogue less intelligible and dramatically increases listening fatigue.   Without clear un-compressed dialogue there can be no intellectual depth or meaning.   The problem is made worse when the background music and sound effects are used to mask the dialogue, causing the sound to appear as a background wash behind the picture.   This loss of intelligibility also forces the listener to interpret the dialogue.   How often we sometimes need to play back repeatedly to understand a word or phrase.   This miss-use of compression and noise masking to dumb-down films, is driven by ignorance and trend, not by educated understanding.

Future opportunity   for cinema sound is when large scale digital projection becomes cost available and the historical Academy characteristic format that applied to celluloid film no longer applies.   With digital projection the sound format can be open architecture similar to the future applications described above for live sound.

For home cinema the compression and EQ per-emphasis should now be encoded separately allowing the user to reduce or remove it on playback.   Also digital management could allow the background music and sound effect levels to be adjusted separately from the dialogue.   The denial of allowing this interactive management at the consumer level is a major limitation for home cinema.

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Seven Basic Principles

The first principal is to make the music come alive.   This also means real, beginning with   no effects,   no EQ,   no compression-limiting,  simply relying on microphone technique, musician and instrument placement, and natural room reflections.   Only from this initial starting position can the correct use of effects, EQ and compression-limiting enable one to create the desired outcome.   This can be looked at in 3 parts.

1 Balance   relation of loudness of each instrument and voice to each other.
2 Image         pan positions (left - center - right) of each instrument and voice.
3 Dimension   managed by EQ and effects, placing instruments and voices forward or back.

The following descriptions assume a basic background of using of mixers, EQ equalization, graphics, parametric, effects, microphones, leads connectors, power amplifiers and speakers.

1. Block Diagram and Flow Charts

Mixing desks have basic features in common but each feature can vary in complexity.   The inputs may   or may not   have phantom power available for microphones with internal pre-amps.   The EQ may be a simple (bass and treble) control similar to domestic sound systems   or   a complex parametric system that allows for any band of frequencies to be selected and adjusted.   The auxiliary sends may   or may not   be switched pre or post EQ,   and or   pre or post fader etc.   Some mixers have 1 or 2 independent stereo input channels whereas other designs require 2 separate channels to be used with an external locking bar for grouping 2 faders.   Mixing consoles in recording, film and TV studios may be modified after purchase to enable a greater range of flexibility.

All mixers are supplied with a block diagram sometimes described as flow charts.   A flow chart or block diagram is not an electronic circuit, but a representation of the circuit layout.   It is essential to understand flow charts to be able to know what functions the mixer has and the order in which the functions are arranged.   Flow charts also show what parts of the mixer can be externally accessed or separated, and which parts are not accessible.

(a) Balanced Inputs   The example flow chart below, of a single channel shows that the input can be selected for a balanced XLR mic or a line level jack plug.   Balanced means that the mic signal is between XLR pins (2 and 3) and are not referenced or common to earth.   This is done to stop electrical noise interference between earth and signal pins from being amplified.   The balanced input circuit only responds to the signal between pins 2 and 3, and does not respond to interference noise which is common to the signal pins and  earth.

The line input (jack) signal is attenuated (reduced) to a lower level similar to the mic level.   In more complex mixer designs the line input has a separate pre-amp.   A separate line level pre-amp is the best option.

A balanced isolation input transformer (TX) stops cable noise and other electronic interference from getting to the first pre-amp.   Many mixers use a electronic circuit to achieve a similar result.   The transformer is best option.

The first pre-amp also changes the input signal from balanced to un-balanced to be processed within the mixer.   The signal is only made balanced again when re-sent from the mixer.

(b) Phantom power 48V   is a technical trick to switch a supply voltage for powered mics to the XLR signal pins (2 and 3).   This saves using extra wires or an external power supply and the reason for describing it as phantom power.   This will be explained in more detail on the mic and cable pages.

(c) Phase switch   The phase invert (Inv) switch is essential to manage different mics to insure they are in-phase when used collectively, eg. a drum kit.   However there are interesting techniques of 2 mics that are used out of phase to create acoustic comb filter effects.   Another technique is for live application where 2 mics are placed on top or beside each other and the vocalist only sings into one mic at close range.   If the mics are out of phase the common background spill will be minimised.

(d) Gain   Each channel has an input gain control.   The pre-amp gain (volume) control increases the small mic signal (approx 10mV) up to line level (approx 1V).   Fully clockwise increases the mic input signal X 100 (+40dB).   This allows the input signal to be adjusted for the main fader to be put into the correct operating position.

(e) -20dB Pad   Mic level is approx 10 - 100mV (1/100 - 1/10 Volt)   Mixers have a -20dB Pad attenuation switch which reduces the incoming signal level to 1/10 (-20dB).   This is required to avoid the first pre-amp from being overloaded (clipping) especially for dynamic mics with very loud singers at close range and mics placed close to drum kits.

Some loud pop singers only perform with the mic at or in their mouth.   Screaming at close range and can cause a dynamic mic to produce 1V which can easily overload and distort the input stage.   A simple recording trick is to allow the singer a mic to scream into at close range that is fed back to their headphones.   The separate recording mic is placed at a correct distance, sometimes without the pop singer being aware.

(f) Hi-Pass Filter HPF   limits low frequencies below 100Hz, to stop vocal popping and bass rumble.   Some mixers allow the hi-pass frequency to be adjusted; this also is the best option.

(g) Insert jack   The signal from the HPF is made available at an insert jack.   This allows the signal to be processed through an external effect unit and then returned to the mixer EQ.   Some mixers have extra inset jacks pre and post EQ and main fader.

(h) EQ   equalization can be a simple bass treble or complex parametric, which allows frequency bands to be selected and adjusted.

(i) Aux   The auxiliary outputs are independent and can be switched pre or post main fader.   Some mixers have many Aux sends.   Some auxiliary sends are dedicated to pre or post fader including pre or post EQ, and some mixers allow the Aux sends to be switched from the different locations.   The latter is essential for foldback which does not require the EQ that is selected for the assign recording output from the main fader.

(j) PFL   The pre fade listen, or solo switch is beside the main fader.   Some mixers enable the PFL function to be switched pre or post EQ.   Some professional mixers have an independent VU or peak meter for each channel.   But most basic mixers only have meters for the outputs.   The PFL switch will connect that channel directly to one of the output meters for monitoring its level.   As the EQ can change the channel level it is essential to check that the PFL is reading the post EQ position for the meter reading to be correct.

(k) Main fader and assign   The fader pre-amp returns the signal to line level at the pan control, then to the 8 output selector switches (left 1,3,5,7)   (right 2,4,6,8).   All bus lines are sent to the output stages of the mixer and are also simple to follow.

Assign management and layout differs on each mixer.   The above channel allows the operator to decide which of the output bus each channel is assigned to and for which purpose it is used.   Some mixer designs assign the pan output to a separate A B or stereo bus.

www.soundcraft.com/mixing/basics
www.sound.westhost.com/mixer/technical

2.  Understanding dB

dB   is how our ears hear sound as a ratio.   (also refer to dB page)

  1.26 X   Watts power change   (1dB)   is the smallest sound change we can hear.
  2       X   Watts power change   (3dB)   is heard as a slight difference.
10       X   Watts power change   (10dB)   is heard as 2 X as loud to our ears.
100     X   Watts power change   (20dB)   is heard as 4 X as loud to our ears.

A microphone converts the sound to a Voltage.   The mixer increases and modifies the signal Voltage.   Increasing the signal Voltage is described as gain.   A mixer functions with signal Voltage but with very little current Amperes.

The power amplifier increases the signal Voltage to a much higher level and adds a large amount of current Amperes to drive the speaker.   Volts X Amperes = Watts (power)   We hear the power as sound from the speaker described in dB.

0dB can be referenced to a any particular Voltage or Watt (power).

The above graph shows a comparison of Watts referenced to 0dB as 1W.

Watts power = (Volts X Amperes)
Increasing 10 X Voltage, increases 10 X Amperes.   10V X 10A = 100W and so on.

When working with mixers and effects we describe the audio signal as its Voltage or gain.
Changing the signal Voltage (gain) changes the loudness power we hear.
Increasing the signal Voltage (gain) X 2 causes the power to be increased X 4.

  2 X Voltage change in the mixer   is       4 X Watts change in power amplifier (6dB).
10 X Voltage change in the mixer   is   100 X Watts change in power amplifier (20dB).

This simply shows that increasing the V X 10 also increases the A X 10.   Therefore the Watts power is increased by 100.   We hear the power increase as a sound change of 20dB.

This is the reason when describing dB referenced to Voltage gain   the dB numbers are X 2 compared to describing dB referenced to Watts power.   By looking closely again at the above graph
(100W is 20dB)   (10V is 20dB)   and so on.
(100W is 20dB)   (100V is 40dB)   and so on.

Understanding dB is simple,   but it takes 4 hours practice to use it;   once learnt never forgotten.
Riding a bush bike is simple,   but it took 4 hours practice to do it;     once learnt never forgotten.

VU  (Volume Units)   and   dBu   (deci Bell units)

Line level   0 VU   is the internal signal level of mixing consoles, signal processors, effects units, and recording hardware and is a reference that allows hardware to be interconnected.   0 VU is verbally pronounced as 'Oh' for 0 and the V U are said as the letters.

0dB (m) (V) (W) etc   means that the   0   is a reference to a stated Voltage or Watt.

0 dBm   =   0 dBu   =   0.775 Volt   (775 milli Volt)   This 0VU reference was adopted in early audio history to represent a signal Voltage of   775 mV RMS   as   'Line Level'.   0dBm refers to 1 milli Watt across 600 Ohms which co-insides with being 775 mV.   This reference applied to valve technology pre 1960 and no longer applies.   This dBm and dBu as 0.775V is confusing and should have been deleted many years ago.

0 VU   should be referenced to 1V as dBV.

Professional audio   O VU   +4dBm (1.23 V)

Professional audio should be 0 VU   =   +6dBV (2 V)   (if I was dictator)

Domestic audio   O VU   is   -10dBV (316mV).   The very low line level of -10dBV (316mV) is the result of portable CD Walkman and MP3 I Pods being powered from 2 x 1.5V batteries (3V).
A 3V supply rail is small therefore the signal level has to be small.

Audiophile   O VU   is a random magical number   (making it difficult to inter-connect equipment).

3.   Signal level

Understanding signal level and dB is essential to obtain the maximum dynamic range from the mixing process.   Without understanding signal level and dB it is not possible to obtain the maximum dynamic range from the mixing process.   Therefore one is forced to use excessive compression to squash the dynamic range to stop everything from being distorted.

The majority of analogue professional mixers and effects have an internal supply voltage of  
(+15V DC) and (-15V DC)   referred to as The Rails.   The total supply voltage between the rails is 30V.   High power professional amplifiers have a rail supply of 80V to 160V.   The test audio signal is a continuous sine wave of   A 440Hz   or   1KHz.

It can be seen in the above pic that the maximum AC music signal is limited by the rails.   The peaks of the sine wave cannot go further than the rail limit of   +15V and -15V.   This is described as
30V Peak to Peak   (30V P/P).

RMS (Root Mean Square) is a mathematical averaging of the AC music over 1 second.   The approx RMS V can be obtained by dividing the P/P / 3   (30V / 3 = 10V RMS).   However the exact formula is (P/P / 2) X 0.707

If the signal is increased the peaks of the sine wave will be clipped by the rails.   Extreme clipping is used by guitar players as sustain.   Sustain simply means sound remaining at the same level, by being clipped.   Valve amplifiers give rounded soft clipping whereas solid state amplifiers clip with precise sharp edges.

Many solid state pre-amps and power-amps have short circuit protection that is momentarily activated by clipping, which can put extreme spikes into the signal.   Many roadie sound engineers deliberately drive PA's into clipping to obtain an aggressive distorted sound for heavy metal music.

6dB Headroom   6dB is double or half of the voltage.   To insure against the possibility of clipping, the audio signal must not go above 1/2 the rail voltage.   The maximum signal at the onset of clipping is 10V RMS therefore the audio signal must not go above 5V RMS.

By paying close attention to the above pic it can be seen that music is very complex with transient peaks.   Depending on the type of instrument the transients can be well over 20dB of the music body.   However transients can be peak limited to within 20dB without noticeable change to the music.   10dB transient peak limiting is often accepted.   10dB transient allowance puts the body of the music at 1/3 below 5V at approx 1.23V RMS (+4dBu)

Transients are so short in duration that they may not be detected by mechanical analogue VU meters.   Solid state display meters can have clipping detect, showing transients with slow decay, including the RMS body of the music.   Many lager mixing consoles have small peak meter readings on each channel, on top or beside the faders.

By paying close attention to the above pic is simple to obtain a dynamic range of 40dB but difficult to obtain a dynamic range of 60dB.   To obtain 60dB dynamic range is a juggling act between the maximum allowable signal level and the composite noise floor.   It is difficult but not impossible to get the collective noise floor below 1mV (-60dBV) which is the minimum to obtain a 60dB dynamic range.   The graph below is a general overview of the juggling act.

Noise floor   Each brand markets is product with claims of virtually zero noise level, often at the theoretical limit of below 100uV, which easily allows for a dynamic range of 90dB.   But the noise floor is limited by component physics which product manufacturers have little to zero control over.   Composite noise is multiplied by the total amount of hardware in the mixing recording and playback chains.

4.   Attenuators   Log / Linear

Attenuation means to stop resist or reduce.   A water tap is an attenuator.   There are stepped attenuators that consist of switches in 1 to 3 dB steps.   Most attenuators are smooth moving controls for gain volume and EQ   (rotary potentiometers and straight faders).

A simple carbon track made by a pencil will conduct electricity and function as a resistor.   A smooth attenuator has a wiper that moves along the carbon track, as in the pic below.   However the materials used in high quality pots and faders are more complex.   By looking at the pic below it can be seen that the signal level comes in at the top of the attenuator.   The wiper selects a smaller signal level allong the track.

Linear   as the word describes 50% of wiper attenuator movement selects 50% of the signal   direct 1:1 attenuation.   Linear is used for EQ and most calibration controls.   Liner enables easy adjustment of signal at the top 50% of the attenuator movement,  whereas it is difficult to manage a small signal level at 10% of the attenuator movement.

Log   logarithmic means the first 50% of fader movement selects 10% - 20% of the signal level   depending on log curve.   Log is used for volume and gain adjustment.   Log allows for easy management of small signal levels,  but log does not allow for easy management of large signal levels at the top of the attenuator movement.   The top 20% of attenuator movement makes a large 50% change to the signal level which is difficult to manage.

The above pic shows a comparison between log and linear faders at 50% movement.   The reference 0dB is at the top of the faders.   The attenuation is written as -dB attenuation down each fader   shown for comparison only.   The faders attenuate the signal level and therefore there is a pre-amp after each fader to re-amplify the signal to 0VU.   The graph below shows the correct reference for the 0dB markings beside the faders.

0dB reference   markings are on all controls of the mixer.   When all the controls are set at the 0dB mark, the signal level that comes into the mixer is the same level that leaves the mixer.   This is described as the default or re-set position.   From the reference of 0dB on the log fader the signal gain can be increased +10dB at the top of the fader.

Repeat   By paying close attention to the above pic it can be seen that a log taper allows for easy manual detailed adjustment at low signal level.   The first 50% of the physical movement from the bottom enables approx 10% (-20dB) of the available signal level to be easily managed.   The top 50% of the fader movement controls approx 90% of the signal level.   This top 50% movement of the fader gives the feeling of being over-reactive (similar to an on/off switch).

Log taper is best suited to the lower 50% of movement for accurate and detail control of signal level.   Log taper faders are intuitive for 50% of recording engineers, whereas   Linear taper faders are intuitive to the other 50% of recording engineers.

Fader design problems   If you pre-adjust the incoming signal level to the fader to have the maximum signal required when the fader is at top position then linear taper will give the best mechanical control for obtaining accuracy and detail.   50% of the downward movement enables 50% (6dB) of the available signal to be managed.

The most important design function missing on mixers is a switch to change the fader taper from Log to Linear.   Linear taper faders require a gain attenuation control to adjust the correct operating signal for the fader.   Linear faders are better suited to live mixing, whereas Log faders are better suited to recording.   The failure of console manufacturers to recognise and provide log/lin fader adjustment is one of the greatest limitations of mixer management.

5.   Panning

Mono   the same sound from both left and right speakers introduces an acoustic comb filter effect which is harmonically pleasing, but reduces high frequency energy, presence and sharpness.   Panning to one side so the sound comes from a single point, increases sharpness and sounds closer.

A clear center image can be created with a stereo mic or 2 mics panned hard left and right in close x y orientation.   Experimentation will find the best result.

Mixing for recording is often done on 2 small speakers at close range (near field).   Speakers that are 4 meters (12ft) or less apart is acceptable for a centre image (mono) with minimal loss of fidelity.   However, for live productions speakers may be 12 meters (36ft) or more apart.   Live mixing in mono, the same sound from left and right speakers at large distances apart introduces a greater comb filter effect and fidelity will noticeably deteriorate.

Facing speakers directly forward also adds excessive reflection from walls and further reduces intelligibility.   Many roadie sound engineers mix in mono in front of only one speaker stack.   The speaker system should be turned inward to improve directivity and minimise wall reflection.   Wherever possible, live mixing should be from the centre in stereo, where sound from left and right intersects,   at a distance no further back than where direct sound from the speakers is equal to the reflected reverberant energy of the room (critical distance).

Spatial movement.   It is important to pay attention to how a pan control functions.   Most domestic pan controls simply reduce the level to the opposite channel to which the knob points to.   However, pan controls on professional mixers lift by +3 the direction the knob points to   as well as reducing the opposite track toward zero.   This allows for even spatial movement during panning.   The majority of music recordings do not require spatial movement from panning.   Live productions do benefit from spatial movement effects.

Spatial movement is essential for Cinema recording.   But as a person or the bass rumble of a space ship (did I say space ship?) moves across the screen, it sometimes sounds wrong.   Console manufacturers fail to provide attenuation adjustment for the pan center point to -6dB for tailoring spatial movement.   Most are fixed at -3db.

Often for a sound image to give a smooth movement from center to left or right, an increase of 6dB is required.   Many skilled recording engineers prefer to use separate channels with automated faders, programmed for obtaining the desired outcome for spatial movement.

6.   EQ Equalization

The primary use of EQ is shelving.   Shelving filters unwanted frequencies from being amplified or recorded.   Eg. Stopping bass frequencies and popping being amplified from vocal microphones.

The secondary use for EQ is tonal colouring and presence.   Presence (approx 2K Hz) can cause an instrument or voice to have the feeling of being positioned forward or back.

For live productions, avoid at all costs the use of graphic parametric and Q controls in extreme boost positions.   This increases the gain of the center frequency, often causing acoustic ringing and instant mic feedback.   It is possible to use high Q in cut positions to control mic feedback problems.   A false belief is that EQ Q in cut positions can reduce room reverberation or compensate for peaks in poor quality speaker systems,  this has no more effect than simply reducing the level.

Q   is a ratio number referring to   Quality of resonance.   Filter networks in graphic and parametric equalisers cut or boost a given center frequency pushing it toward resonance.   If the Q is boosted too high, music information that co-insides with the center frequency will ring.   This ringing reflects complex and discordant harmonics on both sides of the center frequency.   Also the slopes of the filter rotate phase in opposite directions.   Applying Q to a narrow band of frequencies creates distortion, but if applied subtly can have a pleasing effect for managing specific instruments.

To improve clarity and presence, the best practice is to use shelving and minimal slopes over a broad range as shown in the pic below,  especially for live productions.   However, primary EQ management is best applied with appropriate microphone technique.

Slopes   are most effective for rotating the frequency response over a broad range at a gentle angle (3dB / octave) over I decade (ratio 1:10).   This broad EQ management achieves overall brightness and closeness for voice and voice instruments with zero distortion or ringing and maintains full fidelity.   The above pic shows a useful example between 300Hz and 3KHz.   Above and below the knee frequencies, the response is returned to flat.   This is essential to enable frequencies below 300Hz to be heard and to stop frequencies above 3KHz from being unnecessarily boosted.   This rotation angle can also be reversed to reduce harshness or obtain a softer feel to the voice and voice instruments.

Spectral energy   The average spectral energy of music is flat, between 100Hz and 1Khz.   Music has approx equal energy on either side of middle C or 300Hz.   The highest musical notes are approx 2KHz.   Above 1KHz the harmonics decrease in energy at approx -6dB / octave.

At 2KHz the energy is approx -6dB (1/4).  
At 4KHz the energy is approx -12dB (1/16).  
At 10KHz the energy is approx -20dB (1/100).  
Sound systems need higher power for bass speakers, but less power for high frequency tweeters.

Delay

Delay   has a multitude of uses for recording, giving a feeling of thickening a voice or instrument.   A voice or instrument as mono in both channels left and right with one channel delayed, causes an illusion of left or right bias positioning, separate from listening position.   This delay effect can also be enhanced with a stereo mic.

Repetitive diminishing delay is heard as reverb and echo, which is pleasing when used subtly, but reverb and echo reduce intelligibility and forces an association of distance.   Introducing a space between the original sound and the reverberation, limits the reverberation from reducing intelligibility.   For live productions, adding reverb or echo often exaggerates problems in an already overly reverberant room.   (Echo is delay, heard as distinct repeat).

7.   Compression-limiting

When skilfully used, compression can allow subtle sounds to be heard while louder sounds are kept under control.   A whisper and a scream can be kept at the same level.   Also live production can benefit with peak limiting to avoid the sound system from being driven into distortion.

p

The above pic is the simplest limiter that can be made.   An LDR (light dependant resistor) is connected across the input pin of the jack plug to earth.  A small 12V light bulb is connected across the speaker output.   As the light bulb lights up with the music, the LDR resistance decreases and partially shorts out the signal.   The irony is that this simple 2 component circuit works so well that some valve compressor limiters actually use this circuit (with a few more components) and make lots of money.

By paying attention to the graph above, compression can be viewed as;   the high levels above 0VU being compressed to lower level;   or the low levels below 0VU, being increased to higher level.   The result is the same, everything gets squashed together.   This example shows the compressor set to 2:1 ratio.   Peak limiting can be adjusted to any level and changed in ratio and used with or without continuous compression.

Many compressor limiters also include noise gates.   A noise gate turns off the input at very low signal levels so not to hear noise hiss from the mixer circuit or the background studio noise picked up by the microphones.   Noise gates are also used with guitar players that have very noisy valve guitar amps that hum and buzz continuously.

Ducking or Cross limiting   Complex compression techniques are used for TV and film when the main dialogue is used to compress the background music.   The dialogue causes the background music to reduce (duck below) enabling speech articulation to be retained.   The dialogue automatically acts on and compresses the music and background effects in real time.   The overall level remains the same.   This is heard on talk back radio, when the announcer talks over the caller.

Peak limiting   is essential to avoid high level transients from clipping.   Most analogue mixing consoles have a internal supply voltage of   +-15V = 30V P/P   which will allow a maximum signal of   7-10V RMS   20dBm approx.   The allowed standard states that music signals must not be greater than -6dB from the maximum voltage limit.   This is referred to as 6dB headroom.   Allowing for the 6dB headroom and maximum 20dB transient capacity the average analogue mixing console, depending on noise floor, has difficulty in obtaining a 60dB dynamic range.

Transients can be much greater than 20dB ratio 1:10 above than average music level.   Applying peak limiting to control transients to within 20dB of the average music level has almost zero audible effect on the music,  however limiting transients to lower levels within 10dB may have a noticeable effect.

The time the peak limiting takes to act and release should not interfere with the waveform cycle.   This is a compromise as seen in the above pic.   Acting too fast will distort the low frequencies causing excessive non symmetry in the waveform resulting in 2nd harmonic distortion.   Too slow and the first few high frequency transients get through.   When the loud passage ends, the release delay time can cause the music signal to be reduced in level or at worst momentarily reduced to zero.   Peak limiting and compression create distortion and therefore should be used subtly and skilfully.

The above pic shows excessive compression limiting in steps.   Peak limiting is used to reduce the transients to almost zero.   Compression then reduces the dynamic range by squashing the music, reducing the upper level and increasing the lower level,  then the compressed music is increased up in gain to the maximum headroom level.   At this maximum level, with almost zero dynamic range and with all the transients, nuance, harmonics, and detail removed, along with removed stereo depth of field, the sound is at one mono-tonous LOUD level.

Compression Madness   Commercial marketing drives the worst of this behaviour.   Radio and TV broadcasting is restricted within a limited dynamic range.   The aim is to make the pop music or commercial appear to be louder by compressing the music.   This enables the average level to be artificially increased.   The next trick is to add aural exciters which distort the harmonics within the music; similar to fingernails on a school black board.   There are many variations in how this is done.

The above left pic represents a live orchestra.   The above right pic represents the recorded over-compressed orchestra.   Small sounds are brought up↑ and louder instruments are pushed down↓.   The dynamic power of the orchestra is compressed so all instruments are at the same level.   Everything can now be heard on small studio monitors and small domestic sound systems.   The compressed sound gives an illusion of the full orchestra - but this is illusion.   The compressed orchestra is highly inter-modulated and distorted.   Small low fidelity speakers mask this distortion.

When an over-compressed recording is played back on a large full fidelity active sound system, the inter-modulation and distortion caused by excessive compression is now revealed.   Intolerable to listen to, flat and lifeless, the instruments no longer sound natural.   Virtually no nuance, stereo imaging and depth of field remain.

Best practice is to apply compression-limiting to individual voices and instruments   s-e-p-a-r-a-t-e-l-y.   Applying compression-limiting to a completed mix causes the instruments and voices to inter-modulate each other.   Worst of all is applying compression limiting to a compiled stereo mix.   The (left/bar) - (right/bar) out of phase components are exaggerated, resulting in loss of stereo image and depth of field.   Exaggerated examples of this are radio, TV compression and internet MP3 etc.

Other web sites that have excellent information on mixing and the over-use of compression at the bottom of the page.   Please give yourself time to read the pages and go to the links listed on them.

www.cdmasteringservices.com/loudnes graphic
sound.westhost.com/compression.htm

Noise Reduction

Noise reduction,   (pre digital stone age)   was originally required for noisy analogue tape machines, noisy mixing desks, noisy valve processors, and the analogue sound tracks on celluloid film stock which are still in use.   There are 3 basic noise reduction techniques.

(a) Noise gates   simply switch the input off at a low level so the floor noise can not be heard.   This had the unfortunate effect of removing the low signal harmonics and nuance.   Noise gates are commonly used for speech dictaphones court recording etc.   The noise gate can also be used to turn on and off the pause function of a tape or digital recorder so as not to record vacant time.

(b) High frequency pre-emphasis   is used for noisy analogue film stock.   The high frequencies are boosted when recording onto the film stock, and reduced on playback.   Also used for analogue tape recorders.

(c) Companding   The signal is compressed when recorded and then increased to the maximum level the tape will allow.   On playback the signal is expanded and returned to normal.   The residual tape noise is not heard.

These techniques were combined in complex arrangements to achieve the best results.   However during the 70s and 80s many recording engineers claimed all noise reduction techniques interfered with the fidelity of the music.   Some recording engineers went to extraordinary effort to achieve almost zero studio noise without using external noise reduction units.   Dolby and DBX are names commonly associated with noise reduction management and have detailed information on their web sites.

www.soundman.com reducing noise

Monitors

The purpose of a small near field monitoring system is for the engineer to hear an approximation of small domestic sound systems.   This allows the engineer to hear what he/she is doing when applying EQ and compression-limiting.   The studio should have a full fidelity 4-way active monitoring system.   This would enable a separate fully dynamic mix to be released for the audiophile market.

Without the reference of a large full scale active monitor system to hear what the EQ is actually doing, the major problem of inconsistencies between musical recordings will remain.

The extreme of this problem is for cinemas (Academy characteristic) compensating for large low fidelity 2-way cinema speakers and high frequency screen and room attenuation.   When the recorded music is played back on a full fidelity active sound system, the harshness created by excessive EQ (compensating for the cinema requirement) is audibly annoying.

Unfortunately the majority of EQ and compression in pop recording is used to compensate for limitations of the small studio monitors,   often without the recording engineer being aware.   Small 2-way monitors and domestic speaker systems are referenced at 1 Watt with a static tone sweep at close range,   and become distorted as the power increases.   This problem is never mentioned in the technical specs of the monitors.   Most recording engineers experience listening fatigue without understanding this cause.

Many pop recording engineers have a religious belief in marketing hype and a romantic attachment to brand names.   The failure to have a proper understanding of basic electro-acoustic principles (physics of loudspeakers) and the belief that 'technical knowledge' is not relevant   is the primary cause of the problem.

Polar Response   2-way small speaker systems are the majority, and have wide dispersion at low frequencies, and narrow beaming dispersion at high frequencies.   This problem becomes worse as the power increases.

The small woofer
Below middle C   a small bass speaker cannot effectively couple to the air.   As the power increases with deep bass notes, the cone over modulates and flaps at the limit of exertion.     The bass notes do not increase with level and begin to acoustically compress.   The excessive woofer cone exertion causes the mid to be muddled (inter-modulation).

The tweeter
Tweeters do not compress with power and increase in level untill destruction without warning.   8in cone speakers (woofers) become distorted above 1KHz (lobing distortion).   The tweeter forced to be crossed over at a lower frequency than it is designed for, between 2K - 3KHz.   Most tweeters are designed to give correct performance above 5KHz (not 2KHz).   This is not a problem below 1 Watt, where the tweeter is not stressed,  but at higher power many dome tweeters generate extreme 2nd harmonic distortion as a result of the lower crossover frequency.

Inter-modulation   Linearity and Intelligibility.   Muddle-ness and inter-cluttering within the music, difficulty to discern detail,   caused by interference within speaker components.   Lobe and node distortion is caused by secondary vibrations and chaotic resonances within the speaker cone at higher power sounding harsh and screechy.

Dynamic power response   is the ability for the sound fidelity to remain intact between low and high power for dynamic range.   It is not possible for small 2-way monitors or small domestic speakers to achieve this accurately.   The majority of sound systems are passive, due to cost and the fashion for systems to be small.

For a monitor speaker system to sound consistent and accurate at all power levels it must be 3 - 4 way active (with 15in bass speakers).   Each speaker driven by its own amplifier, and matched in efficiency, power and dispersion.

The closing comment for this page is to bring awareness to a recording industry that has historically placated to mixing sound suited to small low fidelity speakers.   This over compressed material should be freely available on the internet reflecting its true value.   For the recording industry to mature and take its place in the digital world, it needs to let go of the past, and make available to the world full fidelity recordings that are best heard on large scale active sound systems.

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Other web sites that have excellent information on mixing.  
Please give yourself the time to read these pages and go to the links listed on them.

Articles on mixing and compression
www.mindspring.com/~mrichter/
www.cdmasteringservices.com/dynamicrange.htm
www.georgegraham.com/compress.html
www.dbxpro.com compression white paper
www.soundonsound.com compression

www.soundonsound terms glossary
www.soundonsound mixing articles
www.rane.com/library
www.behringer.com/tutorials
www.tweakheadz.com
www.obsolete.com

Companies products and organisations
www.filmsound.org
www.rane.com
www.dolby.com
www.dbxpro.com
http://www.mackie.com
www.allen-heath.com
www.soundcraft.com