AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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| Week 4 Equipment 1 |
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Here we go guys.
For some of the more technical mixing desk questions you might want to actually look at some desk manuals that can be found in our downloads sections and also on company websites.
There is a wealth of information available and the idea is to extract some meaning for yourselves. By all means use this forum thread to "discuss" some of the topics that you have trouble with. Discussion is vital so we can start to reflect on our learning as we progress. I'd like to see you all debating a little between yourselves this week.
It's fine to ask each other questions in the threads about the topics.
Part A
1. What is the function of mixing desk automation?
2. Why might the faders have to be moved during the mix?
3. What aspects of mixing would normally be automated on an analogue console?
4. Comment on VCA vs. Moving Fader automation.
5. Describe write/update/read.
6. How does an automated mix session start?
7. How would a section of automation data be re-recorded?
8. How are automation 'punch-ins' blended smoothly with existing data?
9. How could a complex series of moves be increased in level by say 3dB?
10. Describe fader grouping.
11. How could EQ, for instance, be automated using an analogue mixing console?
12. How would EQ be automated using a digital mixing console?
13. What is 'recall'?
14. What advantages does a digital console have over an analogue console regarding recall?
15. Does a compressor act upon loud signals or quiet signals? What does it do to the signals upon which it acts?
18. What is the function of 'make up gain'?
19. How is a compressor usually connected to a mixing console to compress a single signal?
20. How is a compressor connected to a mixing console to compress the entire mix?
21. Explain 'ratio', in the context of compression.
22. Describe the difference between hard knee and soft knee.
23. What are 'breathing' and 'pumping'?
24. What happens if the release time is too long?
25. What does it mean if the gain reduction meter is showing frequent changes?
26. Describe the effect of compression on noise level.
27. Describe the function of the stereo link switch.
28. What is the side chain input?
29. Describe a typical use of the side chain input.
30. Is a noise gate usually effective on a mixed stereo signal?
31. Is a noise gate effective on a signal where the instrument plays all the time?
32. Is a noise gate effective on a single noisy signal where there gaps in the playing?
33. Why is it often considered beneficial to gate some or all of the mics on a drum kit (except the overheads)?
34. How many gates are often used in live sound: none, a few or many?
35. How is the noise gate connected to the mixing console?
36. What is the function of the Threshold control?
37. What is the function of the Range control?
38. Comment on the Attack, Hold and Release (Decay) controls.
39. What would happen if a stereo signal was gated, but the stereo link function was not selected?
40. Why are side-chain filters beneficial?
41. Comment on the use of an external key to improve the reliability of gating.
42. What is an expander?
43. Why are compressors and gates often used together?
44. Comment on envelope shaping using a noise gate.
45. Comment on gated reverb.
Part B
Pre Fade and Post Fade busses can be found on most modest desks. Outline for what purpose you might use each type of buss for a recording session and also a mixing session. Outline what you feel to be the difference in approach to the two.
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Last edited by AUdIoCoUrSeS on Mon Mar 29, 2004 9:56 pm; edited 1 time in total |
Mon Mar 29, 2004 9:16 pm |
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iNSTiNCT2765
Joined: 05 Nov 2003
Posts: 60
Location: Denmark |
hey Chris, check question 16 8O
- Aman |
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Mon Mar 29, 2004 9:46 pm |
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hoggs33
Joined: 09 Feb 2004
Posts: 55
Location: Nottingham, England |
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Part A
1. What is the function of mixing desk automation?
Automation enables you to record the movement of faders and effects changes that are then ‘applied’ to the mix automatically when the track is next played back. This enable complicated mixes to be created that may otherwise be physically impossible to recreate unless several pairs of hands are available which may lead to overcrowding etc. Automation can be ‘real-time/dynamic automation’ or ‘scene/snapshot automation’. Real –time records the fader, panning, effects etc in real time to each track as the song plays. Snapshop involves saving the mixer data at intervals rather than throughout the entire song. It helps also to avoid mistakes when mixing down to stereo.
2. Why might the faders have to be moved during the mix?
To adjust the levels of each instrument in a certain part of a mix to create a better dynamic mix, or one section may have been recorded too loudly and needs slight adjustment only part way through a song to achieve and maintain a good balance. It may also be necessary to apply effects in only a certain section of a song. Instruments may be added to the song as it progresses or when the vocals start so subtle adjusting of the faders may be necessary to avoid too much ’clutter’ in the mix or clipping.
3. What aspects of mixing would normally be automated on an analogue console?
Faders, panning, effects settings, EQ etc.
4. Comment on VCA vs. Moving Fader automation.
VCA is Voltage Controlled Amplifier which is an amplifier that will change gain according to the level of control voltage sent to it. VCA Automation is therefore a system of computer control of channel gain (or other functions) by use of voltage controlled amplifiers that change gain according to the level of control voltages sent to them by the computer. The faders do not actually move with VCA as all the level control is going on out of sight in the VCA’s. With moving fader automation, the computer actually controls the faders which move during playback. VCA’s have an update or trim mode where you can superimpose one set of moves on top of another so you won’t overwrite the automation. These systems therefore have the benefit of being better for updating since there can be total independence between fader position and audio level. Updating is more complex in moving fader systems. Moving fader systems will generally give a better sound quality. Power cost automation systems tend to use VCA’s as the level controlling element. Whilst VCA’s can be high quality these days, it needs to be remembered that unless a separate preamp is used for recording, the signal will pass through a VCA at least twice in the multitracking process, multiplying any deficiencies.
5. Describe write/update/read.
Write is the actual ‘writing’ of the automation data. ‘Update’ as referred to above is the making of any adjustments to the original ‘write’. ‘Read’ is what is read when the song is played back i.e the actual reading of the data as it is applied to the mix. These are the three modes of automation that the desk can be set to depending on what is required.
6. How does an automated mix session start?
The simplest way to automate a mix is to use scenes or snapshots. The faders would be set in positions that suit the various sections of the song, and record, or ‘write’ each as a snap shot, which would then be replayed by the computer in sync with the timecode on the tape. The alternative would be to set rough fader positions which will be good for most of the song and write these into the computer. Then you can set about writing moves for the individual faders, based upon the starting positions.
7. How would a section of automation data be re-recorded?
As referred to above a VCA system can update easily using the update or ‘trim’ mode which superimposes one set of moves on top of another. To trim you would set the fader to halfway and upward moves will increase the level, downward moves will decrease it. Alternatively, you may have to set the fader to a ‘null’ position where its physical position matches its audio position, as indicated by a pair of LEDs, but the principle is the same. VCA automation systems are great for updating since there can be total independence between fader position and audio level. In the case of moving fader systems, updating is a little more complex since the level of the audio is inextricably linked to the position of the fader knob. Moving fader systems usually have touch sensitive fader caps so if you want to change a move, all you have to do is grab the fader and do it. The computer will interpret this as a temporary switch into write mode for as long as you touch the cap.
8. How are automation 'punch-ins' blended smoothly with existing data?
By using crossfades
9. How could a complex series of moves be increased in level by say 3dB?
On some systems such as Sonar 3 that I use there is an option to process the audio and increase or decrease the levels by a set amount of 3dB, so I would use that!!!!! If that is not available then the faders for the series of moves could be routed to an aux bus and the fader of that bus then increase by 3dB.
10. Describe fader grouping.
This is where a group of faders are routed to one bus. For example if a drum kit is recorded over several tracks such as one track for kick, one for snare etc. if these are grouped together and routed to one fader on the separate bus then it enables the overall level of the drums to be controlled using one fader rather than having to adjust several faders. Once the individual drum sounds are mixed correctly you then only have to adjust one fader to set the overall level of the drums in a mix.
11. How could EQ, for instance, be automated using an analogue mixing console?
Using write, read and update?
12. How would EQ be automated using a digital mixing console?
By using the write function and adjusting the EQ as necessary. Then use read and update if required.
13. What is 'recall'?
When a snapshot setting is taken, recall can be used to access previously saved settings.
14. What advantages does a digital console have over an analogue console regarding recall?
15. Does a compressor act upon loud signals or quiet signals? What does it do to the signals upon which it acts?
A compressor acts on both loud and quiet signals and reduces the dynamic range. It can be used to make quiet passages louder by using gain and louder passages quieter by compressing them.
18. What is the function of 'make up gain'?
To bring the compressed signal back up to the original level.
19. How is a compressor usually connected to a mixing console to compress a single signal?
It would be connected directly to the appropriate channel strip and is known as a line or insert effect. The compressor is placed in the signal path so that all the instrument’s sound passes through the effect.
20. How is a compressor connected to a mixing console to compress the entire mix?
In this instance the compressor would be connected to an aux bus and the mix then routed through this bus.
21. Explain 'ratio', in the context of compression.
The ratio is the amount that the compressor affects the signal. A ratio for example of 2:1 means that for every decibel that the signal goes over the threshold setting, it is reduced by two. In other words, if a signal goes 1dB over the threshold setting, its output from the compressor will be only ˝ dB louder. At ratios of 10:1 the compressor will start to act like a limiter.
22. Describe the difference between hard knee and soft knee.
Most compressors give the option of choosing between hard knee or soft knee. Hard knee and soft knee refer to how the compressor behaves as the input signal passes the threshold. Hard knee applies the compression at an even rate regardless of the level present over the threshold. So, if you choose a compression setting of 4:1, the compressor applies this ratio for any signal over the threshold limit. Hard knee compression is used for instruments like drums, where you need to clamp down on any transients quickly.
Soft knee applies compression at a varying rate depending on the amount the signal is over the threshold setting. A soft knee setting gradually increases the ratio of the compression as the signal crosses the threshold until it hits the level that has been set. Soft knee compression is used on vocals and other instruments where the signal doe not have fast peaks.
23. What are 'breathing' and 'pumping'?
When a compressor is making large changes to the input signal (10 to 12dB or more), the noise floor will also rise and fall with the signal level. When this noise signal rises and falls drastically between signals, such as on a heavily compressed, noisy drum track, you may hear the noise level ‘breathing’ between drum hits. One solution is to turn up the release time so that the noise floor won’t have time to rise between drum hits. However if the release time is too long, lower level signals after the peak will be lost as the compressor slowly stops reducing again. This is called pumping, as the lower level signal (noise included) slowly fade back up to their normal signal level.
24. What happens if the release time is too long?
Pumping, as referred to above in question 23.
25. What does it mean if the gain reduction meter is showing frequent changes?
Basically that the compressor is doing its job effectively and compressing the signal.
26. Describe the effect of compression on noise level.
The maximum gain of compression occurs when the input signal falls below the threshold, which can sometimes lead to the boosting of background noise during quieter passages or pauses. Also noise level can be increased if the compressor is set too high. If set too high the loud portions are being compressed enough to make the level of the softest sections of the music, including any noise much louder in comparison. In order to eliminate this problem and get rid of the noise the ratio or threshold settings should be turned down.
27. Describe the function of the stereo link switch.
Most two channel compressors have a stereo link switch. If this is activated then stereo signals can be processed. In link mode both channels are compressed by the same amount, thus preventing the image from moving to one side or the other when a loud sound appears on only one side of the mix.
28. What is the side chain input?
The Side Chain allows the signal passing through the unit to be controlled by the dynamics of another separate signal. Connection to the side chain is usually made via the rear panel jack sockets.
29. Describe a typical use of the side chain input.
To ‘de-esss’ vocal – i.e get rid of sibilance. The sibilants can be selectively removed by compressing only when there is excessive level of high frequencies.
30. Is a noise gate usually effective on a mixed stereo signal?
No noise gates are usually only effective on single unmixed signals.
31. Is a noise gate effective on a signal where the instrument plays all the time?
No, because the sound of the instrument will drown out any background noise – noise gates are only effective when there are periods of silence.
32. Is a noise gate effective on a single noisy signal where there gaps in the playing?
Yes, this is the best use of a noise gate, for example when recording a guitar, there may be noise from the amp when there are section when the guitar is not playing – the gate can be used to cut out this background noise.
33. Why is it often considered beneficial to gate some or all of the mics on a drum kit (except the overheads)?
To cut or reduce the spill from other mics
34. How many gates are often used in live sound: none, a few or many?
35. How is the noise gate connected to the mixing console?
As a noise gate can only process one unmixed signal, the place to connect it is in the channel insert point of the mixing control. Gate are not used via the aux send and return loop.
36. What is the function of the Threshold control?
The threshold sets the level in dB at which the gate opens (i.e stops filtering the section). The gate allows all signals above the threshold setting to be passed through unaffected, whereas any signals below the threshold setting are reduced by the amount set by the range control.
37. What is the function of the Range control?
The range control is similar to the ratio control setting on a compressor except you chose the amount (in dB) that you want the signal to be reduced (attenuated) by the gate. For example a setting of 40dB drops any signal below the threshold setting by 40 decibels.
38. Comment on the Attack, Hold and Release (Decay) controls.
The attack sets the rate at which the gate open in milliseconds. Fast attacks work well for instruments with fast attacks such as drums, whereas slow attacks are better suited to instruments with slow attack, like vocal.
The hold setting controls the amount of time that the gate stays open after the signal drops below the threshold. Once the hold time is reached, the gate closes abruptly. This parameter is listed in milliseconds. – Phil Collins in the 80’s used this quite a lot on his drum sounds.
The release setting dictates the rate at which the gate closes after the signal hits the threshold again listed in millisecond. Unlike the hold feature, the release setting does not close abruptly, rather it slowly (according to the release setting) close. This produces a more natural sound. The release time should ideally be set so that it matches the natural decay time of the instrument, otherwise you can get a clipped-off sound.
39. What would happen if a stereo signal was gated, but the stereo link function was not selected?
The channels would change state at slightly different times resulting in a very poor sound.
40. Why are side-chain filters beneficial?
If a noise is so high that setting the threshold to a point that fits neatly between the signal and the noise is impossible, a solution would be to use a side chain filter. The side chain is used to control the behaviour of the processed signal and in a gate is commonly called the trigger, or key signal.
41. Comment on the use of an external key to improve the reliability of gating.
If problems still persist that are not remedied by using a side chain filter then another alternative is to try an external key to open the gate. This is set up by feeding the signal from a regular mic through the gate as normal. A contact mic is then taped to the instrument or amp and is fed via a preamp to the external key input of the gate. The gate should then be switched to external and the threshold set to a reliable setting. The gate is then only triggered by a signal that picks up virtually no external sound only the sound of instrument being treated.
42. What is an expander?
The expander is to the gate what a compressor is to a limiter – instead of reducing the volume of notes below the set threshold by a specified amount, it reduces them by a ratio. In other words, with the gate you set a certain amount indB, that a signal is reduced, and with the expander, you reduce the signal by setting a ratio. The ratio changes the signal gradually, making the affected signals sound more natural. An expander would be used when you want to subtly reduce noise from a track rather than just filtering it out completely, for example when dealing with breath from a singer. If a gate is used, you get an unnatural-sounding track because the breaths are filtered out completely. Ith the expander, you can set it to reduce the breath just enough to be not so noticeable, but you can leave a little of the sound in so that the singer sounds ‘normal’
43. Why are compressors and gates often used together?
Compressors always have the effect of increasing noise level as the peaks of the signal are brought down in level, bringing them closer to the noise floor. Make up gain is then applied to bring the overall signal level back up again, raising the noise floor at the same time. A noise gate is then used in conjunction with the compressor to remove the noise.
44. Comment on envelope shaping using a noise gate.
A noise gate can also be used as a creative tool as well as a problem solver as they can also be capable of a variety of envelope shaping effects. The classic use is to put a more or less continuous signal through the noise gate such as a drum loop or distorted chords from an electric guitar, and then use the external key to chop it up into a rhythm. This is used quite a bit in dance music and also rock music – the guitar hook for example in ‘What’s the Frequency Kenneth’ by REM.
Another useful gate effect is to compress the sound of an individual drum, then gate it. This works particularly well on drum samples which have a little bit of reverb on them. The compressor can shape the envelope of the sound by emphasizing the attack (by setting a slow attack time on the compressor, allowing the initial transient to get through unaltered), or by allowing the reverb to increase in level as the drum dies away. The noise gate can then further process the envelope using the attack, hold and release controls.
45. Comment on gated reverb.
This is to gate the sound very heavily to give it a sharp cut off point when the reverb is still loud – Phil Collins again I’m afraid!!!!
It would be set up as follows on say a snare drum. The mic from the drum would be connected to the console. The through an auxiliary send, send some of the signal to a reverb unit. Bring the output of the reverb back to a channel with an insert point. Connect the noise gate to the insert point send and return of the reverb channel. Connect the insert send of the snare channel to the gates external key input. Set the hold and release controls so that the reverb extends beyond the end of the dry snare drum sound, but then dies away suddenly by using a long hold and short release. Hey Presto you are Phil Collins in the mid 80’s!!!!!!
Part B
Pre Fade and Post Fade busses can be found on most modest desks. Outline for what purpose you might use each type of buss for a recording session and also a mixing session. Outline what you feel to be the difference in approach to the two.
Pre Fade and Post Fade busses are included on a mixing console primarily to allow the user to set up foldback mixes or add effects from external effects units. However there can be several novel uses of these busses by using a little imagination.
Foldback is of course needed so that he performers can hear any tracks already recorded etc. Foldback is set up using the pre fade send control, which feeds some of a channel signal into a mono mix buss running the length of the mixer and out via an Aux master level control. A foldback signal is referred to as pre fade because it is picked up before (i.e pre) they reach the channel fader. The significance of this is that, once set, the level of the signal does not change if the channel setting is varied. This then eliminates having to adjust any foldback mixes every time a channel fader may be moved. If a console has more than one pre fade send then independent foldback mixes can be created to meet the requirements of each performer.
Post fade busses are the opposite in that they pick up the signal feed after (post) the channel fader, so any change to the channel fader position will also affect the Aux send level. This is what is needed if the Aux send is being used to feed an effects device - as the channel fader setting is modified during the course of a mix, the amount of effect needs to be changed by the same amount to maintain the correct proportion of effect to dry signal. It is possible to send different amounts of each channel’s signal to the same effects unit by using different settings of the post fade send control on each mixer channel. Utilising the Aux send system has the advantage that different amounts of the same effect can be added to different instruments in a mix. For example, one reverb unit could be used to provide a rich reverberation treatment for vocals, less for drums and a little or none for guitars and bass.
A pre fade mix is useful if using a desk that has fewer output groups than required, because you can use any pre-fade send output to feed a multitrack recorder input. In other words the pre-fade send is being used as an additional output group. The levels would of course have to be controlled using the knob rather than the faders but otherwise it would be the same.
Post fades could also be used as tape outputs, providing the mixer has routing buttons.
Both post and pre fade can be used for a variety of purposes to get more out of your console by using a little imagination |
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Fri Apr 02, 2004 6:35 pm |
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iNSTiNCT2765
Joined: 05 Nov 2003
Posts: 60
Location: Denmark |
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Week 4 – Equipment 1
Part A
1. What is the function of mixing desk automation?
The function of mixing desk automation is to perform fader movements automatically whenever the mix is played back. The fader movements produced by whoever is mixing are stored in a computer and then reproduced during playback. This is extremely useful because it saves time since it virtually eliminates the need for getting everything exactly right on one take. Without the automation, if the mix called for gain riding and such, it would all have to be done manually and on one take. Automation eliminates the need to do the entire mix over if something is not sitting right in it because the movements are stored and played back
2. Why might the faders have to be moved during the mix?
At different stages of a song, the vocals may need to be turned down to make room for more instruments for example, in the chorus and this would have to be done by moving the faders. Some instruments may also want to be faded in and out at particular times during the course of the song.
3. What aspects of mixing would normally be automated on an analogue console?
Besides automating the volume by moving the faders, some analogue consoles can also automate channel mutes and effects send/return mutes. When nothing is playing on the channel, it can be muted to eliminate background noise.
4. Comment on VCA vs. Moving Fader automation.
Moving Fader automation is controlled by a computer, which stores the fader movements and then reproduces them when the track is played back. The audio signal is routed through the fader and the faders move along with the automation during playback, which means that the fader level always relates to the actual gain setting of the different channels and gives you a visual of that. Moving Fader automation is however expensive and the physical mechanical faders can be noisy, causing distraction while playback of a mix.
VCA’s are Voltage Controlled Amplifiers. No audio passes through these faders but when moved, they produce a DC voltage, which is then read by an automation computer that then determines the fader position. A drawback is that during playback, the faders don’t actually move so you have no visual of the gain setting for each channel unless the system can be connected to a VGA monitor. But since the faders don’t move during playback, they can’t cause any distraction sonically.
5. Describe write/update/read.
In the world of mix automation, write is the term used for recording the fader movements. Update is when you overwrite parts of the automation with new automation to fine-tune the mix. Read is the playback of the automation.
6. How does an automated mix session start?
You pick the part of the song and the tracks that are to be automated. Then if the automation desk is in write mode, the data generated by the fader movements should get stored in the computer.
7. How would a section of automation data be re-recorded?
If the automation needs to be fine-tuned then the automation desk is simply put into write mode and the fader movement produced overwrites the previous. This is also called an update…right?
8. How are automation 'punch-ins' blended smoothly with existing data?
Using moving faders the fader is already set to the level corresponding to the exact gain so a punch-in doesn’t cause a level jump. But using a VCA system, the fader has to be set to the corresponding level to avoid a level jump when updating to punching in new automation. If a VGA screen is connected to the system, it would most likely show the automation gain of the channels plus the actual fader levels. This can be used to match up the faders to avoid the level jumps.
9. How could a complex series of moves be increased in level by say 3dB?
By grouping the channels on which the moves are produced. Then one fader can be used to control the collective gain of all those faders.
10. Describe fader grouping.
When the collective gain needs to be raised or turned down on many faders, instead of doing it individually, the faders can be grouped and one fader can then control the collective gain.
11. How could EQ, for instance, be automated using an analogue mixing console?
Have the same audio signal routed to a couple of channels and make different EQ settings for each. Then by moving the faders you can cross fade between the different channels and record the movements to automation.
12. How would EQ be automated using a digital mixing console?
EQ can be automated using digital consoles by saving different EQ settings to different scenes or memories. Then the automation can be used to switch between the different scenes.
13. What is 'recall'?
A major difference between analogue and digital consoles is that with digital consoles you have the ability to save settings such as EQ, panning, effects, dynamics, fader levels and even routing settings into scenes. Different settings can be made and saved to separate scene memories and instantly recalled when needed.
14. What advantages does a digital console have over an analogue console regarding recall?
Think I answered this one in 13.
15. Does a compressor act upon loud signals or quiet signals? What does it do to the signals upon which it acts?
A compressor is used to maintain the incoming audio signal by decreasing the dynamic range of the signal. It reacts if the signal gets too loud by turning it down by a suggested ratio and it reacts on quiet signals by turning them up. This decreases the dynamic range and gives a more even audio signal.
18. What is the function of 'make up gain'?
The ‘make up gain’ is used to turn the audio signal back up after it has been compressed, because compression lowers the input signal.
19. How is a compressor usually connected to a mixing console to compress a single signal?
A single compressor is usually connected to the mixer as an insert. This means it is connected between the audio source and the mixer.
20. How is a compressor connected to a mixing console to compress the entire mix?
To compress the whole mix, a stereo compressor can be connected to the main outputs of the mixer. This is done again as an insert but this time it is between the mixer and the recorder. Using my digital mixer, I route all the channels I want to compress to the Bus Out, which is connected to the compressor and in turn, the compressor is connected to a couple of inputs on the mixer. Then I can compress the stereo mix and record it back into the computer.
21. Explain 'ratio', in the context of compression.
The ratio sets the amount of compression that is applied to the signal when it exceeds the threshold. For example, if the ratio is set to 4:1, then for every 4 dB that exceeds the threshold level at the input stage, only 1 dB is output before the make up gain is set.
22. Describe the difference between hard knee and soft knee.
Hard knee compression is compression that is applied at full as soon as a signal exceeds the threshold level. Soft knee compression is compression that is gradually applied to the signal at a low ratio that is slowly increasing to the set ratio as the audio signal is reaching the threshold level. When the threshold level is reached and exceeded, the compression should be at full level. Because of this, soft knee compressors work in a subtler manner on the audio signal.
23. What are 'breathing' and 'pumping'?
If the release time is set too short, the gain will come up quickly during the silent bits and the level of the noise floor will be increased. This is called breathing. You can hear this on solo vocals during the pauses when the singer breathes. The breaths are loud and completely audible because the gain increases quickly due to the short release.
Pumping can be heard best on an overall mix. If one instrument is louder than the others, it will be the main trigger for the compressor. If that instrument then stops playing, the rest of the mix will be increase in level. Then when the instrument is again introduced, the rest of the mix will be turned down again.
24. What happens if the release time is too long?
The compressor slowly stops reducing gain and the lower level signals slowly come back up to their normal levels.
25. What does it mean if the gain reduction meter is showing frequent changes?
The gain reduction meter shows the compression that is being applied to the input signal. If it shows frequent change it means that the signal that is being input is exceeding the threshold a lot.
26. Describe the effect of compression on noise level.
Using compression brings up the noise level or noise floor because while the loudest bits are being compressed, they are being turned down. Then when the ‘make up gain’ is applied, everything is being raised in level, including the noise floor.
27. Describe the function of the stereo link switch.
The stereo link switch on a dual mono compressor, links the two channels together. Then compression can be applied to a stereo signal with the settings being controlled by only one of the channels (usually the left one).
28. What is the side chain input?
The side chain input is a separate input on the compressor that allows the input signal to be compressed according to the dynamics of the signal connected to the side chain input instead of it’s own dynamics.
29. Describe a typical use of the side chain input.
A typical use of the side chain input is de-essing vocals. This is done after the vocals are recorded and the ‘sss’ or the sibilance needs to be controlled. You could EQ the vocal track but that could possibly dull the track when no sibilance is present. De-essing with the side chain is like an EQ that only steps into play when the sibilance is present. To do this, you duplicate the vocal track so you have two copies. One is routed to the input of the compressor and the other signal is routed to the side chain input. Boost the high frequencies in the sibilant area (around 7Khz) on the vocal track that is routed to the side chain. Now whenever there is an excess of sibilance, the compressor will react and reduce the gain during the sibilance.
30. Is a noise gate usually effective on a mixed stereo signal?
No, a noise gate would not be effective on a mixed stereo signal because, firstly it won’t work properly if it is a dense mix with a lot going on. Secondly, why would you need a noise gate on a mixed stereo signal? The noise gate is used to close the signal after a specified time to eliminate noise when recording your audio to hard disk or tape. So your stereo mixed shouldn’t even need a noise gate.
31. Is a noise gate effective on a signal where the instrument plays all the time?
Not if you want the instrument to play all the time.
32. Is a noise gate effective on a single noisy signal where there gaps in the playing?
Yes, and this is what it is most commonly used on.
33. Why is it often considered beneficial to gate some or all of the mics on a drum kit (except the overheads)?
This is beneficial because it gates the input when no signal is passing through and this eliminates the possibility of spills on the separate tracks. The overheads aren’t gated because they capture the entire kit and the cymbals so the audio signal is not meant to be closed at any time.
34. How many gates are often used in live sound: none, a few or many?
Probably depends on the instruments and such but ATLEAST a few. Don’t really have too much experience with live sound so I actually have no clue.
35. How is the noise gate connected to the mixing console?
It is connected as an insert just like the compressor.
36. What is the function of the Threshold control?
The threshold on a noise gate determines the level at which the gate closes. If the signal ducks below the set threshold, the gate will close.
37. What is the function of the Range control?
The Range controls the attenuation of the gated signal.
38. Comment on the Attack, Hold and Release (Decay) controls.
The Attack controls the rate at which the gate opens when the signal exceeds the threshold level.
The Hold controls the amount of time the gate is left open after the signal dips under the set threshold. The Release determines the rate at which the gate closes when the signal drops under the threshold.
39. What would happen if a stereo signal was gated, but the stereo link function was not selected?
If the stereo link function on a gate is not selected, the left and right channel of the stereo signal will be gated individually. If the two gate channels have different settings, one channel of the signal may be gated quicker than the other. Use the stereo link when working with stereo files.
40. Why are side-chain filters beneficial?
Side chain filters can be useful when gating because they can be used to specify certain frequencies that open the gate while other frequencies won’t affect the gating.
41. Comment on the use of an external key to improve the reliability of gating.
An external source can be used to trigger the gate. This is useful because it eliminates the possibility that the gate opens by accident.
42. What is an expander?
This is the opposite of a compressor. While the compressor crushes the dynamic range of an audio signal, an expander expands the range. The quiet parts get quieter and the loud parts get louder.
43. Why are compressors and gates often used together?
When using a compressor, the noise level is also brought up with the rest of the signal with the ‘make up gain’ so a gate is used to close the signal while there are pauses such as when recording a kick drum.
44. Comment on envelope shaping using a noise gate.
A noise gate can be used for envelope shaping. A continuous loop can be feed into the noise gate and triggered externally by a drum loop to create a nice rhythm effect.
45. Comment on gated reverb.
This is another envelope shaping effect that can be achieved using a noise gate. It was popularised in the 1980’s. It basically means that the reverb just suddenly cuts off at some point instead of fading out. This is also an effect that has been emulated and can be found in most reverb effects modules.
Part B
Pre Fade and Post Fade busses can be found on most modest desks. Outline for what purpose you might use each type of buss for a recording session and also a mixing session. Outline what you feel to be the difference in approach to the two.
The pre fade bus can be used to feed a foldback mix to the artist or musician during the recording session. The reason a pre fade bus is best for this is that moving the fader positions don’t affect the mix in the fold back. This means that the foldback mix doesn’t have to be mixed over if you adjust the recording volume for the instruments. A post fade bus is good for feeding effects modules because the amount of effect should be relative to the sound so if you add some reverb to a snare drum and then decide to turn down the snare a little bit, the reverb should also turn down automatically. During a mixing session, a post fade bus is also useful to feed a recorder because after setting the levels, they can be recorded into the recorder as they sound. |
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Sat Apr 03, 2004 12:05 am |
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griff505
Joined: 23 Feb 2004
Posts: 68
Location: Bristol |
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Part A
1. What is the function of mixing desk automation?
The most common form of mixer automation is as a means of storing fader positions dynamically for reiteration at a later point in time. This assists the engineer during mixdown when the number of faders that need to be handled at once becomes too great for one person. Automation data is digital and stored in a computer built into the mixer or in a computer connected to a mixer. DAW’s allow for eq, effects and pan position to be automated as well as fader level.
2. Why might the faders have to be moved during the mix?
To adjust the level of, or mute certain tracks during a mix. For example a solo violin in an orchestra will need to have its volume raised for its solo section, mix automation enables this. Eq may need to be adjusted on certain instruments when other tracks start, for example certain frequencies in an orchestra may need to be reduces when a vocal track starts. The pan position can be moved to apply special effects similar to a Leslie speaker.
3. What aspects of mixing would normally be automated on an analogue console?
Volume and mute.
4. Comment on VCA vs. Moving Fader automation.
VCA – The position of the fader is stored and the data is used to control the gain of a VCA (Voltage Controlled Amplifier) or digital attenuator. The faders themselves do not physically move, therefore the fader’s position may not always correspond to the gain of the channel. It is possible however to combine the two approaches whereby gain control can be performed be a VCA and the fader being moved mechanically to display the gain. This approach allows rapid changes in level and for dynamic gain offsets of a stored mix whilst retaining the previous gain profile.
An automation system would be added in a break point between a fader and the corresponding VCA. The automation processor reads a digital value corresponding to the position of the fader and can return a value to the VCA to control the gain of the channel. The fader position is measured by and analogue-to-digital converter which turns the DC value from the fader into a binary number which the microprocessor can read. The automation system scans the fader position many times a second and reads their values.
The disadvantage of the system is that it is not easy to know what the level of the channel is as during the read process the automation computer is in charge of the channel gain, rather than the fader. There are ways around this however; a mixer’s bar-graph meters can be used to display the value of the DC control voltage which is being fed from the automation to the VCA, or a separate display can be provided for the automation computer including fader position and channel gain.
Moving Fader – Works in a similar way to VCA automation with the addition that the data returned to the fader is used to set the position of a drive mechanism which physically moved the fader to the position which it was at when the mix was written. This gives the advantage that the fader always represents the gain of the channel. Clutches are used to remove the danger of a fight between the fader and engineer in a situation where the engineer and the automation system both attempt to move the fader, and the fader is usually made touch-sensitive to detect the presence of a hand on it.
5. Describe write/update/read.
Write – VCA gain corresponds directly to the fader position
Update – VCA gain controlled by a combination of previously stored mix data and current fader position
Read – VCA data controlled by data derived from a previously stored mix.
6. How does an automated mix session start?
The section of the track that needs to be automated will be cues up, the mixer should be set to write mode and the necessary moves planned out. The tracks should be played and the moves made. The mixer can then be set to read mode and the tracks can be listened back to ensure the desired movements were made, if they are not perfect they can be altered using update mode.
7. How would a section of automation data be re-recorded?
Using VCA automation update mode involves using the relative position of the fader to modify the stored data. The absolute position of the fader is not important as the system assumes that the starting position is a point of unity gain, only changes to the faders position are added to the stored data. Therefore in update mode it a fader is increased by 3dB the overall level of the updates passage would be increased by 3dB.
Moving Fader automation would allow the engineer to re-record the fader position through the use of a clutch activated by the touch-sensitive fader.
8. How are automation 'punch-ins' blended smoothly with existing data?
Punch-ins are not an issue with moving fader automation as the faders will be set to the correct level by the built in motors. However when using VCA automation punch-ins can be a problem as the automation computer is in charge of the channel gain, rather than the fader. Therefore VCA fader are commonly provided with null LEDs, which are little lights on the fader package which point in the direction that the fader must be moved to make its position correspond to the gain of the VCA. When the lights go out, or are both on the position is correct. Most systems will only switch from read to write when the null point is crossed to ensure a smooth transition.
9. How could a complex series of moves be increased in level by say 3dB?
Moving Fader Automation usually has some form of relative mode which can be used to offset a complete section by a certain amount. The problem with this is that if there is a sudden change in the stored mix data while the engineer is holding the fader, it will not be executed. The engineer must let go for the system to take control again. (This is where the combination of moving fader and VCA-type control is preferable).
Using VCA automation the channels which need to 3dB increase would be best routed to a group channel which could be used to increase the gain of all the channels in one move.
10. Describe fader grouping.
Faders can be grouped to reduce the number of faders that need to be moved during a mixdown. This is usually achieved using dedicated master faders. The automation computer may allow any fader to be designated a group master for a group of faders assigned to it. The user can set up a fader as a group master; the level from this fader will then be used to modify the data sent back to all the other VCA’s in the group, taking into account their individual positions.
11. How could EQ, for instance, be automated using an analogue mixing console?
By routing multiple signals of the same audio to a number of channels and setting the desired EQ for each channel. The faders would then be automated to blend between the different channels, favouring the channel with the desired EQ at the correct point during the track.
12. How would EQ be automated using a digital mixing console?
On a digital mixer EQ can be automated precisely using digital signal processing (as well as delay, phase, routing, delay, reverb, compression, etc).
13. What is 'recall'?
Recall is a feature on digital mixer which allows snapshots of all settings of the mixer to be stored and recalled with the pressing of a button or a MIDI Program Change Command.
14. What advantages does a digital console have over an analogue console regarding recall?
It is only fader levels that can be automated on an analogue console. Moving fader automation allows for fader levels to be recalled when set to read mode, however VCA automation recalls the fader levels, but does not move the faders to the set position.
15. Does a compressor act upon loud signals or quiet signals? What does it do to the signals upon which it acts?
A compressor is like an automatic volume control, when the audio is loud it gets turned down and when it is soft it gets turned up. This means sharp signals are now curved and fading signals are now picked up and last longer. It also means smoother sounds and fatter notes.
18. What is the function of 'make up gain'?
The compressor reduces the level of any signal which exceeds its threshold; therefore the output will tend to be lower than the input. An extra stage of make-up gain is used in order that the output level can be matched to any subsequent piece of equipment.
19. How is a compressor usually connected to a mixing console to compress a single signal?
Compressors are connected in line with a signal, and when compressing an individual signal the channel insert points should be used.
Compressors are not generally connected via the auxiliary send system although some engineers do use this method when using analogue mixers as adding the compressed sound to the uncompressed sound can give musically useful results. A compressor is a processor not an effect so attempting this on a digital or software based system will probably result in nasty, phasey sounds due to the slight time delay between the two signal paths.
20. How is a compressor connected to a mixing console to compress the entire mix?
When compressing the entire mix the compressor should be patched to the console’s master stereo insert points, although if there are none fitted the compressor can be connected directly between the mixer outputs and the mastering device’s inputs, and the mastering device used to monitor the compressed sound and control any fades.
21. Explain 'ratio', in the context of compression.
Ratio = The change in output level that results from a change in input level, i.e. 2:1 – a 2dB change in input level will result in only a 1dB change in output level. (A ratio of 10:1 or over is know as limiting where an input exceeding the threshold is subjected to a high level of gain reduction, and the output is prevented to raising about the threshold be a significant degree).
22. Describe the difference between hard knee and soft knee.
Hard Knee – Works on the threshold principle – signals lower than the set threshold remain unaffected, while those exceeding the threshold are reduced in level in an amount depending on the ratio level.
Soft Knee – 1. The compression ratio is extremely low for small signals and automatically increased as the signal level increases. This is an easy system to set up as there is no threshold or ratio to set, only a compression amount control.
2. There will be a threshold, but the compression ratio increases progressively as the signal approaches the threshold. By the time the full threshold level is reached the full compression ratio is in force.
Therefore soft knee compression obtains a gentler transition between signals that are not compressed and higher-level signals that are compressed, although may not exercise such firm gain control.
23. What are 'breathing' and 'pumping'?
Breathing – A compressor can make large changes to the input signal, up to 10dB or more, which means the noise floor will also rise and fall with the signal level. When the signal rises and falls drastically between signals, such as a heavily compressed drum track, noise level ‘breathing’ may be heard between drum hits. It can be overcome by carefully adjusting the release time of the compressor, whereby the noise floor will not have time to rise between drum hits.
Pumping – If the release time is too long, lower level signals after the peak will be lost as the compressor slowly stops reducing gain. This creates a ‘pumping’ effect as the lower level signals (noise included) slowly fade back up to their normal signal level. Therefore a balanced release time on the input signal is essential.
24. What happens if the release time is too long?
The gain may not have recovered sufficiently by the time the next quiet sound comes along, which will be suppressed more than is necessary.
25. What does it mean if the gain reduction meter is showing frequent changes?
Frequent changes of the gain reduction meter shows that the audio input is frequently exceeding the threshold and compression is being applied.
26. Describe the effect of compression on noise level.
Noise is a problem when compressing singals orininating from noisy sources such as analogue Tep, guitar amplifiers or some older synths, as the noise will be increased by the compressor during quieter sections or pauses. Therefore compression is added to vocal track recorded to tape to ensure healthy recording levels at all times.
Every dB of compression added is a dB of deterioration in the signal-to-noise ratio of the signal passing through it. Therefore compression added to produce 10dB of gain reduction will increase noise in quieter sections by 10dB.
27. Describe the function of the stereo link switch.
A stereo link switch can be found on two-channel compressors and can be used where compression of a stereo signal is required. This allows both channels to undergo exactly the same amount of gain reduction at all times regardless of whether the biggest peaks are in the left or right channel. Linking combines the side chain signals so that compression only responds to an average of two channel levels, and often the compression can only be adjusted using only the controls of the first channel level, or sometimes by averaging the control settings of the two channels.
28. What is the side chain input?
The side chain is the part of a compressors circuitry that monitors the level of the signal being processed. An insert point is often fitted to enable other processors to be connected into the side chain path, or to allow the side chain to be fed from a different source altogether.
29. Describe a typical use of the side chain input.
For example, an equaliser could be connected and signals in the 5 – 8 kHz range boosted. This would allow the compressor to respond to loud, bright sounds in the set range, and can be used to de-ess vocal tracks.
30. Is a noise gate usually effective on a mixed stereo signal?
A mixed stereo signal is usually difficult to treat as there are very few periods of silence. A noise gate works more effectively on a single track which has periods of silence.
31. Is a noise gate effective on a signal where the instrument plays all the time?
No, a noise gate is only effective if there are pauses between then phrases played on an instrument.
32. Is a noise gate effective on a single noisy signal where there gaps in the playing?
Yes, the gate would be set to allow the playing through, but mute the noise in-between phrases.
33. Why is it considered beneficial to gate some or all of the mics on a drum kit (except the overheads)?
If drums are recorded in a live room then the trick of gating reverb is used where reverb is natural in origin. The gate may be triggered via its side chain from the dry drum sound and the ambience mics passed through the gate, usually in stereo. Using a hold time of up to half a second and a fast attack time, a hard-gated drum sound is produced.
A gate can also be used on drum mics to reduce acoustic crosstalk between the mics located in close proximity.
34. How many gates are often used in live sound: none, a few or many?
A few.
35. How is the noise gate connected to the mixing console?
As gate are processors they are always placed in line with the signal being treated, never by the aux send or return loop. The signal may be gated during recording (if a number of signals are being recorded to one track) or during mixing (where any noise accumulated during the recording process can also be gated, and there is no risk of recording an incorrectly gated signal).
36. What is the function of the Threshold control?
The threshold control the level in dB at which the gate opens. Any signal above the threshold passes through untreated, and any signal below it is passed through at a level controlled by the range setting.
37. What is the function of the Range control?
The range control allows a certain amount of signal to pass through even when the gate is closed, e.g. a setting of 10 dB would mean that when the gate was closed, the signal would be reduced by 10dB rather than muted completely. This could be used, for example, to mute the room ambience of a drum recording, or street noise in a film soundtrack.
38. Comment on the Attack, Hold and Release (Decay) controls.
Attack – Determines how long it takes for the gate to fully open once the input signal exceeds the threshold.
Release – This allows the sound to be faded out over a period of time once the signal falls below the threshold.
Hold – Hold time stops the gate entering it release phase for a predetermined time after the input falls below the threshold. In combination with a fast release setting this can be used to create the gated drum sound.
39. What would happen if a stereo signal was gated, but the stereo link function was not selected?
The left and right channels will be gated individually; this would result in poor quality sound as the gates open and close independently of each other. A stereo link function is necessary to prevent the image shifting which occurs if both channels operate independently.
40. Why are side-chain filters beneficial?
The gate’s side chain circuitry measures the level of the incoming signal and compares it with the threshold set by the user, when exceeded the circuit generates a control signal to open the gate at a rate set by the attack control, when the signal falls below the threshold the gate closes according to the settings of the hold and release controls.
41. Comment on the use of an external key to improve the reliability of gating.
An external signal can be used to drive the side chain, allowing one signal to gate another. A set up often used it a bass drum set up to let the bass guitar through and improve the timing off the track. A bass guitar played early will not be let through, and its decay will be at a rate set on the noise gate. On dance track keyboard pads or rhythm guitars are controlled by gates triggered by rhythmic signals such as a drum machine to create a rhythmic chopping effect.
42. What is an expander?
Some gates use the expander principle, which uses the opposite mechanism of a compressor. When a signal falls below a set threshold it is subjected to gain reduction, not total muting. The further the signal falls below the threshold the more gain is reduced. Expanding sounds less obvious than simple gating, and is effective if the threshold is set just about the noise floor.
43. Why are compressors and gates often used together?
A compressor cannot differentiate between a small wanted signal and low-level unwanted noise, therefore noise is set to maximum gain by compressors during quiet passages such as gaps between words and phrases in a vocal track. Many manufacturers now incorporate simple gates / expanders into their compressors to mute the signal at such times.
44. Comment on envelope shaping using a noise gate.
A gate can be triggered via its key input and set with fast attack and release times to create a chopping effect. The key input is sourced from a drum machine or sequenced MIDI instrument with a constant envelope, programmed to play a simple rhythm. If a completely different sound is put into the gate’s main input it will be chopped up according to the key input’s rhythm. A further effect can be created by leaving the original sound untreated, but chopping its reverb, creating a rhythmic feature.
45. Comment on gated reverb.
On a snare drum this would be set up as follows:
- Connect the snare drum mic to the mixing console in the normal way.
- Through an auxiliary send, send some of the signal to a reverb unit.
- Bring the output of the reverb back to a channel with an insert point. (If your console's auxiliary returns have insert points, then they will work fine).
- Connect the noise gate to the insert point send and return of the reverb channel.
- Connect the insert send of the snare channel to the gate's external key input. (You could alternatively derive this signal from another auxiliary send). Set the gate to external key (EXT).
- Set the hold and release controls so that the reverb extends beyond the end of the dry snare drum sound, but then dies away suddenly (long hold/short release).
Part B
Pre Fade and Post Fade busses can be found on most modest desks. Outline for what purpose you might use each type of buss for a recording session and also a mixing session. Outline what you feel to be the difference in approach to the two.
Pre-Fade Send – This is basically another level control feeding a separate mono mix bus that runs across the mixer to the aux 1 master level control and then to the aux 1 output socket. The signal is taken before the channel fader, therefore once it is set it does not change if the channel fader setting is adjusted. The mix set up by the aux 1 controls can provide a monitor mix to musicians exactly to their liking without compromising the main stereo mix. In a studio the mix would be sent to a headphone amplifier distribution box so that multiple performers can listen in. Larger consoles have multiple pre-fade sends so that different performers can be sent different headphone mixes.
Post-Fade Send – This send, aux 2, sends a signal after is has gone through the channel fader. It is used to feed an effect such as reverb, allowing the unaffected signal to pass through the fader then onto the stereo mix bus and the effect unit output is added to this later in the signal path. As the channel fader setting is changed the effect also changes by the same amount, maintaining the balance of dry signal regardless of fader setting. |
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Wed Apr 07, 2004 5:40 pm |
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albertom
Joined: 21 Jan 2004
Posts: 22
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Part A
1. What is the function of mixing desk automation?
The function of mixing desk automation is to perform fader movements automatically whenever the mix is played back. Saves time since it eliminates the need for getting everything exactly right on one take. This eliminates the need to do the entire mix over if something is not sitting right in it because the movements are stored and played back
2. Why might the faders have to be moved during the mix?
In order to obtain the best recording sound, faders have to be monitored or moved. Vocals may need to be turned down or up, same as instruments in order to achieve what you want.
3. What aspects of mixing would normally be automated on an analogue console?
Faders, panning, effects settings, EQ etc.
4. Comment on VCA vs. Moving Fader automation.
VCA’s are Voltage Controlled Amplifiers. No audio passes through these faders but when moved, they produce a DC voltage, which is then read by an automation computer that then determines the fader position. A drawback is that during playback, the faders don’t actually move so you have no visual of the gain setting for each channel unless the system can be connected to a VGA monitor. But since the faders don’t move during playback, they can’t cause any distraction sonically. . VCA’s have an update or trim mode where you can superimpose one set of moves on top of another so you won’t overwrite the automation. These systems therefore have the benefit of being better for updating since there can be total independence between fader position and audio level. Updating is more complex in moving fader systems.
Moving Fader automation is controlled by a computer, which stores the fader movements and then reproduces them when the track is played back. The audio signal is routed through the fader and the faders move along with the automation during playback, which means that the fader level always relates to the actual gain setting of the different channels and gives you a visual of that.
5. Describe write/update/read.
Write is the actual writing of the automation data. Update are adjustments of what was written originally. Read is what is read when the song is played back.
6. How does an automated mix session start?
A default snapshot must be stored set up your static mix and store in the snapshots menu.
The simplest way to automate a mix is to use scenes or snapshots. Select part of the song and the tracks that are to be automated. Then faders must be set in positions that suit the various sections of the song. Then the automation desk in write mode, the data generated by the fader movements get stored in the computer.
7. How would a section of automation data be re-recorded?
Put into write mode and the fader movement produced overwrites the previous.
8. How are automation 'punch-ins' blended smoothly with existing data?
Using crossfades.
9. How could a complex series of moves be increased in level by say 3dB?
Its possible to group the channels that you want to move simultaneously. This is useful because it leaves you with a few empty channels.
10. Describe fader grouping.
Group faders, which is great for changing several controller values at once. For instance, say you have three synths that play the same part, but with different sounds (giving you a big layered sound). You could use a fader group to change all their volumes together, instead of dragging three separate faders every time.
11. How could EQ, for instance, be automated using an analogue mixing console?
Have the same audio signal routed to a couple of channels and make different EQ settings for each. Then by moving the faders you can cross fade between the different channels and record the movements to automation.
12. How would EQ be automated using a digital mixing console?
Change the EQ settings during playback in automation write mode.
Use scenes/snapshots.
13. What is 'recall'?
Recall is reading in the settings that have previously been stored in the memory.
14. What advantages does a digital console have over an analogue console regarding recall?
In a digital console you can have different copies of a track saved on a hard disk, that can be used and eq’d, stuff that are almost impossible to do in analogue recorders.
15. Does a compressor act upon loud signals or quiet signals? What does it do to the signals upon which it acts?
In both. Applying compression reduces the highest levels, reducing the dynamic range. Because the peak level of the signal is now lower, make-up gain is added to restore the original peak level The result is a much more controlled and usable sound.
18. What is the function of 'make up gain'?
Gain Make-Up restores the level lost in the compression process. Since the compressor works by bringing down peak levels, the level of the output signal would be lower than the input if nothing were done. Sufficient gain make up should be applied so that the peaks of the compressed signal are the same level as the peaks of the inputs signal. The sections of the input signal that were quiet will now be louder.
19. How is a compressor usually connected to a mixing console to compress a single signal?
Compressors work at line level, therefore the input signal has to be taken from the mixing console, preferably from the channel insert point send. The output from the compressor is brought back to the channel insert return. By connecting the compressor at this position in the signal chain, its operation is unaffected by the use of any of the console controls, except input gain.
20. How is a compressor connected to a mixing console to compress the entire mix?
Connecting the compressor to the group insert point of the console, or the main stereo output's insert point. In either of these situations, a mix of signals is compressed.
21. Explain 'ratio', in the context of compression.
Ratio is the 'strength' of compression above the threshold level. The higher the ratio, the greater the effect. If the ratio is set at 5:1, it means that when the signal is above the threshold level, when the input signal rises by 5 dB, the output signal rises by 1 dB.
At a compression ratio of 2:1, the effect is mild and suitable for the subtle compression of vocals or for a complete mix. At 10:1, compression is much stronger and more noticeable. Ratios between 5:1 and 15:1 are suitable for the 'compressed' sound, used as an effect in its own right. Higher ratios are used for the control of extremely peaky signals. Above 20:1, the compression effect is so pronounced that it is known as 'limiting'. It is possible to buy a dedicated limiter.
22. Describe the difference between hard knee and soft knee.
A gentler-sounding compression can be achieved by using a so-called soft-knee compressor, where the compression ratio increases gradually as the signal approaches the threshold. Once the signal passes the threshold, the full ratio as set by the user is applied, but, because some compression is applied to signals approaching the threshold, the transition from no gain reduction to full gain reduction is far smoother.
the hard-knee compressor provides firmer gain control, so if a signal is varying in level to an excessive degree, a soft-knee compressor might not produce the required degree of levelling.
23. What are 'breathing' and 'pumping'?
If you over compress either a track or the whole mix you'll notice some rather strange sound effects starting to happen. Too high a compression ratio combined with a low threshold will produce very strange breathing and wheezing type effects on the track.
This "breathing and pumping" occurs when the track has so much compression applied to it that you hear every piece of noise and background hiss coming to the foreground in the gaps in the music. In between the gaps you'll hear just highly attenuated music. The combined effect isn't very pleasant. Even with just high ratios of compression you can start to notice the onset of this effect in parts where the sound changes in volume quickly. Remember the Attack and Release controls.
24. What happens if the release time is too long?
If the Release time is too long, lower level signals after the peak will be lost as the compressor
slowly stops reducing gain. This is called “pumping” as the lower level signals (noise included) slowly
fade back up to their normal signal level. The secret to avoiding these problems is to achieve a balanced
release time on the input signal.
25. What does it mean if the gain reduction meter is showing frequent changes?
Basically that the compressor is doing its job effectively and compressing the signal.
26. Describe the effect of compression on noise level.
Many instruments do not have the sustain that the musician desires, and this can be corrected by using a compressor to extend the note. As the signal fades, the compressor increases its gain, so the note lasts longer.
Another reason is to restrict the dynamic range. By reducing the dynamic range, both can be accommodated at levels that are appropriate, but limited to an acceptable maximum and minimum loudness
27. Describe the function of the stereo link switch.
When a stereo signal is compressed, the stereo link has to be activated so that both channels provide the same amount of gain reduction. If this is not done, a loud signal in one channel will cause that channel to be lowered in level while the other stays the same. Any signal that is panned center in the mix wiill swing in the stereo image towards the unaltered channel. With stereo link selected, the stereo image is maintained.
28. What is the side chain input?
In normal use, the amount of compression or expansion is related to the dynamics of the input signal. The side chain allows the signal passing through the unit to be controlled by the dynamics of another separate signal.
29. Describe a typical use of the side chain input.
Its often used to direct a high frequency boosted signal to the side chain to perform a crude type of de-essing. But applying EQ to the side chain in general, rather than this one specific application allows the compressor that's in your rack right now to have an incredible range of sounds going far beyond the normal differences between models, when used in the standard configuration. You will find that the compressor becomes another type of EQ, but instead of simply cutting or boosting different frequencies, you allow different frequency bands to control the amount of compression applied. When you are in search of that elusive ‘phat’ sound that simple EQ and compression does not give, EQing the side chain is very usefull in this cases. All serious compressors should have side chain EQ built in.
30. Is a noise gate usually effective on a mixed stereo signal?
No. The Noise gate plugin silences sections of a recording which are purely noise. This is an effect which is often used on old films to make the sound appear better without applying drastic noise reduction to the soundtrack. Anything below a specified noise floor is muted entirely. This should be viewed as an alternative to noise reduction and is not usually used alongside it.
31. Is a noise gate effective on a signal where the instrument plays all the time?
No. Think I covered it on question 30.
32. Is a noise gate effective on a single noisy signal where there gaps in the playing?
Yes. As explained on question 30.
33. Why is it often considered beneficial to gate some or all of the mics on a drum kit (except the
overheads)?
For capturing say a clean drumbeat using a microphone you will need to set the threshold by ear, the attack needs to be quick and the hold/release needs to be as short as possible.
34. How many gates are often used in live sound: none, a few or many?
The classical example of where the noise gate would be appropriate is the electric guitar, where the amplifier is likely to be noisy. When the guitarist plays, the sound of the instrument drowns out the noise so there is no problem. When the guitarist isn't playing however, the noise of the amplifier becomes apparent, and irritating. In this situation, what the noise gate does is detect when the signal level is high, when it assumes that the guitar is playing, and opens fully to allow the signal through unimpeded. When the gate detects that the signal level is low, it assumes that the guitarist is not playing and closes completely, blocking off the noise.
Another classic use of the noise gate is on a drum kit, where several might be used. The conventional method of miking up a drum kit demands a mic on each drum, a mic on the hihat and two overhead mics. The problem with this is that the mics are all so close together that each mic picks up every instrument of the kit to an extent, as well as its own instrument. This inevitably blurs the sound. To make it more focussed, all the mics except the overheads are gated so that each channel is only open when the drum is actually sounding. In this situation it is often subjectively better to set the gate so that it attenuates when closed, by perhaps 10 dB, rather than cut off completely. In live sound, this technique is often extended to virtually every microphone for the entire band.
35. How is the noise gate connected to the mixing console?
Since the noise gate can only process one unmixed signal, the place to connect it is in the channel insert point of the mixing console. As with equalizers and compressors, gates are not used via the aux send and return loop. If the console you are working with does not have a patchbay, you will have to make up a special adapter (Y) lead if your console has the usual single stereo jack send/return insert point.
36. What is the function of the Threshold control?
Threshold sets the level above which compression takes place. Signals below the threshold will remain unaltered.
37. What is the function of the Range control?
The Range control sets the degree of attenuation when the gate is closed. Noise gates are commonly set to maximum attenuation unless there is a good reason to do otherwise. On a single signal, the gating effect will be obvious, but of course it should not be so in the context of the entire mix. If the opening and closing of the gate is still noticeable, then the range control should be set to achieve the best compromise.
38. Comment on the Attack, Hold and Release (Decay) controls.
Attack sets the time the compressor takes to respond once the threshold has been exceeded. Attack may be set so that the initial transient of the instrument passes through unaltered, or set to a faster value so that the very start of the sound is compressed. Particularly with drum sounds, careful adjustment of attack time can make the sound more 'punchy' and 'driving'.
Release time plays a very important role in compression. During periods of high signal level, gain is reduced. When the signal level falls below the threshold, the gain will increase at a rate determined by the Release control. If the release time is short, the gain will rise quickly. A long release time will mean that the gain will stay at its reduced level, only recovering gradually:
The Hold control sets a time period during which the gate will remain fully open, even though the signal has just dropped below the threshold.
39. What would happen if a stereo signal was gated, but the stereo link function was not selected?
If the stereo link function on a gate is not selected, the left and right channel of the stereo signal will be gated individually.
40. Why are side-chain filters beneficial?
The side chain allows the signal passing through the unit to be controlled by the dynamics of another separate signal
41. Comment on the use of an external key to improve the reliability of gating.
If the gate still isn't opening and closing reliably. The answer is to use an external key signal to open the gate. Here's the scenario:
Feed the signal from the regular mic through the gate as normal.
Tape a contact mic to the shell of the snare drum.
Feed it via a preamp to the external key input of the gate.
Switch the gate to external key ('EXT').
Set the threshold etc. for reliable triggering.
Now, the gate is triggered by a signal that picks up virtually no external sound. It has to be said that this is probably an over-elaborate technique for most circumstances. But it works very reliably and is worth knowing about for the occasional difficult situation.
42. What is an expander?
An expander is a more sophisticated form of gate. Whereas a gate is either on or off, an expander increases the dynamic range when the signal is below the threshold.
The expander has a ratio control like a compressor. 1:1 means no expansion; 20:1 means that the expander is working almost like a gate.
43. Why are compressors and gates often used together?
Because compression always has the effect of increasing the noise level. The obvious answer is to use a noise gate to remove the noise, providing the signal meets the criteria for gating as outlined above.
44. Comment on envelope shaping using a noise gate.
The noise gate is also capable of a variety of envelope shaping effects, and is a highly creative tool as well as a problem solver. The classic trick is to put a more-or-less continuous signal through the gate, such as heavily distorted chords from an electric guitar, and then use the external key to chop it up into a rhythm. Like this:
Connect the guitar, through a distortion unit, to the gate in the normal way.
Connect a drum machine, or other rhythmic source synchronized to the track, to the external key input.
Switch the gate to external key.
Set the threshold so that the gate triggers on a signal from the external key.
Adjust the attack, hold and release controls to achieve the desired envelope.
Powerful though MIDI sequencing may be, you can't get the same sound in any other way. This is well worth trying.
Another useful gate effect is to compress the sound of an individual drum, then gate it. This works particularly well on drum samples which have a little bit of reverb on them. The compressor can shape the envelope of the sound by emphasizing the attack (by setting a slow attack time on the compressor, allowing the initial transient to get through unaltered), or by allowing the reverb to increase in level as the drum dies away. The noise gate can then further process the envelope using the attack, hold and release controls.
45. Comment on gated reverb.
First popularized in the 1980s, gated reverb has become something of a cliche. But as a technique, it is still well worth knowing about. It goes like this:
Connect the snare drum mic (say) to the mixing console in the normal way.
Through an auxiliary send, send some of the signal to a reverb unit.
Bring the output of the reverb back to a channel with an insert point. (If your console's auxiliary returns have insert points, then they will work fine).
Connect the noise gate to the insert point send and return of the reverb channel.
Connect the insert send of the snare channel to the gate's external key input. (You could alternatively derive this signal from another auxiliary send). Set the gate to external key (EXT).
Set the hold and release controls so that the reverb extends beyond the end of the dry snare drum sound, but then dies away suddenly (long hold/short release).
You could use a distant mic as the reverb source, as an alternative to the reverb unit.
Part B
Pre Fade and Post Fade busses can be found on most modest desks. Outline for what purpose you might use each type of buss for a recording session and also a mixing session. Outline what you feel to be the difference in approach to the two.
When overdubbing, a cue or foldback mix is generally needed so that the performers can hear any tracks already recorded. This is set up using a pre-fade send control, a knob found in the mixer's channel strip which feeds some of the channel signal onto a mono mix buss running the length of the mixer and out via an Aux master level control. The output from Aux 1, for example, feeds onto the Aux 1 buss and then to the Aux 1 output socket, via the Aux 1 master output level control. Foldback or cue signals are referred to as 'pre-fade' because they are picked up before they reach the channel fader. The significance of this is that, once set, the level of the Aux 1 signal doesn't change if the channel fader setting is varied.
It follows that an independent mono mix of all your channels can be set up using the Aux 1 controls, and this will appear at the Aux 1 output where it may be fed to a headphone amplifier or other monitoring system. If your mixer has more than one pre-fade send, you can set up a number of different monitor mixes to satisfy the requirements of each musician, providing you have access to a multi-channel headphone amplifier system. A typical situation is where the backing vocalists want a lot of lead vocal in the cans, whereas the drummer and bass player want to hear primarily each other.
Post-fade Aux send controls pick up their signal feed after the channel fader, so any change to the channel fader position will also affect the Aux send level. This is exactly what we need if the Aux send is being used to feed an effects device, such as a reverb or echo as the channel fader setting is modified during the course of a mix, the amount of effect needs to change by the same amount to maintain the correct proportion of effect to dry signal. By using different settings of the post-fade send control on each mixer channel, it is possible to send different amounts of each channel's signal to the same effects unit.
Utilizing the Aux send system has the advantage that different amounts of the same effect can be added to different instruments in a mix. A typical example might be where one reverb unit is used to provide a rich reverberation treatment for the vocals, less reverb for the drums, and little or none for the guitars and bass.
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