AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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| Week 6 - Equipment 3 |
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Here we go for week 6, looking forward to your responses.
Best of luck.
1. What is the function of automation?
2. Why would faders have to be moved during the mix?
3. What aspects of mixing are normally 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?
8. How would a section of automation data be re-recorded?
9. How are automation 'punch-ins' blended smoothly with existing data?
10. How could a complex series of moves be increased in level by say 3dB?
11. Describe fader grouping.
12. How could EQ, for instance, be automated using an analogue mixing console?
14. How would EQ be automated using a digital mixing console?
15. What is 'recall'?
16. Why is recall necessary?
17. What advantages does a digital console have over an analogue console regarding recall?
18. Describe how you would use a compressor with a side chain input as a de-esser.
19. Describe how you could use a two-channel compressor with a stereo link switch as a de-esser without using the side chain inputs.
20. Explain why a noise gate would not be used when recording Scottish bagpipes.
21. What noise is often heard from an orchestral harp, besides the sound of the strings?
22. Describe how you would use a noise gate to impose a rhythm onto a sustained synth string sound.
23. Describe why the two channels of a stereo compressor must be linked when processing a stereo signal.
24. When mixing a popular music recording using an automated console. If the console is neither digital nor digitally controlled analogue, what facilities would you expect to be able to automate? 25. Often a multitrack recording contains unwanted sounds when instruments are not meant to be playing. State how this can be rectified.
26. On a VCA automated mixing console a number of level changes have been made to the vocal. The producer wishes to keep these changes but increase theoverall vocal level by 3dB. Explain how this is achieved. 27. Is it possible to have an automation system which has moving faders yet the gain is altered by VCAs?
28. Describe the Decca tree system of miking.
29. Describe why `update' or `trim' mode is desirable in a mixing console automation system.
30. State the meaning of EACH of the following sets of initials
i) VU ii) PPM. Briefly describe the difference between VU and PPM meters.
31. With reference to an analogue multi-track recorder briefly explain a) the term `sync' output b) why the `sync' output is required when overdubbing.
32. Describe how a noise gate can be used to create gated reverb on a snare drum, using two microphones and the natural reverberation of the studio.
33. Briefly explain, in terms of quality, the effect of mixing a signal with a version of the same signal delayed by 1 millisecond.
34. State the effect that would be achieved by continuously modulating the delay time up and down. In reference to question 33.
35. Comment on the use of walking surfaces in audio drama production.
36. Why is it not suitable to connect the output of a record player pickup cartridge directly to a studio mixing console? Give two reasons.
37. If it is desired to equalise or compress an entire stereo mix, to which inputs/outputs of the console should the equaliser or compressor be connected? Assume a fully specified mixing console.
38. Describe the function of the stereo link switch in a compressor.
39. What is VCA Grouping?
40. Explain flanging.
41. Exlain chorusing.
42. How can chorus be created with two tape machines.
43. Where does the term flanging come from?
44. What is noise reduction?
46. List some milestones in noise reduction technology.
47. Describe what is meant by plate reverb and spring reverb.
48. Explain the term "in-phase".
_________________
It's all in the ears. - Learn the concepts not the software.
Audio Courses is a way into the music business for you
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Mon Oct 06, 2003 9:17 pm |
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mommi
Joined: 21 Apr 2003
Posts: 47
Location: Tallinn, Estonia |
| Week 6 answers |
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Hi all,
I was happy to see a nice amount or repetition in this week questions. Sure I still feel somewhat unclear about some stuff, automation for example, so looking forward to read any comments on it.
<b>1. What is the function of automation?</b>
Automation allows to control fader operations and/or use of EQ/effects/processors during the mixdown stage. The operations are saved during automation data <i>write</i>. The actual mixdown process then reads the data back in sync with the music.
<b>2. Why would faders have to be moved during the mix?</b>
To achieve proper balance between different instruments/voices and effects.
<b>3. What aspects of mixing are normally automated on an analogue console?</b>
Channel muting and faders. 
<b>4. Comment on VCA vs. Moving Fader automation.</b>
On VCA controlled automation there is no audio signal passing through the fader - the fader just establishes the driving voltage of the voltage controlled amplifier. Moving fader automation utilises motors to actually move the faders during automated mixdown. The audio signal passes though the fader.
<b>5. Describe write/update/read.</b>
Automation data <i>write</i> mode is used for recording the automation data. <i>Update</i> mode is used for later adjustments to it. During the automated mixdown process the automation data is <I>read</i>, ie what you have recorded becomes effective in sync with the music.
<b>6. How does an automated mix session start?</b>
Automation relies on reference timecode. The timecode can be provided by the central timecode word clock or stored to one track of the multitrack recorder. For automation data recording the faders etc have to be set up to their initial position, automation write mode turned on and the multitrack put to play back. Fader movements made during the write mode will be recorded by the console. For the actual mixdown, the initial position of faders etc should be restored and automation set to read mode.
<b>8. How would a section of automation data be re-recorded?</b>
Seems the system has to be in read/write mode. Play back the music, any changes you make will overwrite the already existing data. As I get this, the update/trim mode is cumulative in that it does not destroy the existing data. Would just minor improvements rather than complete overwrite be desirable, update/trim mode should be used. Is it like that
<b>9. How are automation 'punch-ins' blended smoothly with existing data?</b>
By adjusting the system <i>glide time</i>, ie the time it takes to switch from one setting to another.
<b>10. How could a complex series of moves be increased in level by say 3dB?</b>
Group the affected fader(s) under one <i>master</i> fader. Record additional automation data (at that case, 3 db increase) for the master fader.
On a sequencer it may be possible to select the necessary automated region and just apply level rise to it.
<b>11. Describe fader grouping.</b>
Fader grouping is a technique to superimpose movements of one fader to another fader (or other faders). One fader is chosen as being <i>group master</i>. Whatever is done with the master fader will be transformed to the slave faders.
<b>12. How could EQ, for instance, be automated using an analogue mixing console?</b>
Using two or more channels to represent one signal. The channels would be applied different EQ settigns. Switching between the channels by automated muting or fading results in different EQ becoming effective.
<b>14. How would EQ be automated using a digital mixing console?</b>
Change the EQ settings during playback in automation write mode.
Use scenes/snapshots, which is essentially the same.
<b>15. What is 'recall'?</b>
Recall is reading in the settings that have previously been stored in the memory.
<b>16. Why is recall necessary?</b>
To quickly set the console in the necessary state.
<b>17. What advantages does a digital console have over an analogue console regarding recall?</b>
Digital consoles usually offer to store various settings, including EQ, dynamics and effects parameters to scenes or snapshots. These can be read back when needed. There can be a lot of those snapshots.
I’m yet to see an analogue console featuring any store/recall ability at all. If there exist any, there must be much less you can store/recall. Somebody please comment on that.
<b>18. Describe how you would use a compressor with a side chain input as a de-esser.</b>
I would insert the compressor into the vocal channel and also route the vocals through a band-pass filter. I would adjust the filter to pass and emphasise only the hissing frequencies (typically around 7 kHz or so) and direct the filter output to the compressor’s side chain input. This way whenever there are those high frequencies present, the compressor’s dynamic processor will diminish the gain of the signal.
<b>19. Describe how you could use a two-channel compressor with a stereo link switch as a de-esser without using the side chain inputs.</b>
A stereo link switch makes the two compressors behave identically - when gain is reducted in one channel, this is also done on another. For de-essing then I would send the filtered signal to one and the “main” signal to the second compressor. I would use a ratio of 1:1 on the second compressor since I don’t want it to initiate any gain reduction on its own. Now when the first compressor detects a signal (from the filter) it will react by reducing the gain. That behaviour is transferred to the second compressor, where the gain will be reduced by the same amount. All this assumes the stereo link switch is turned on, of course.
<b>20. Explain why a noise gate would not be used when recording Scottish bagpipes.</b>
It is characteristic to Scottish bagpipes to emit a continuous sound (bourdon) through <i>drones</i> even when there may be gaps in melody. Noise gates, on the other hand, are effective in suppressing background noise when no useful signal is present. Since we always have a sound coming out from those instruments, there is no sense in noise gating.
<b>21. What noise is often heard from an orchestral harp, besides the sound of the strings?</b>
Pedal noise.
<b>22. Describe how you would use a noise gate to impose a rhythm onto a sustained synth string sound.</b>
I would insert the noise gate in the synth string channel and connect a rhythm machine to the external key input of the gate. This way whenever the machine fires a signal, the gate will open and pass the synt sound through. When the output of the rhythm machine gets quiet, the gate will close.
<b>23. Describe why the two channels of a stereo compressor must be linked when processing a stereo signal.</b>
Linking makes the two channels behave identically. It means, when there is a compressing action triggered in one channel, that same amount of compression will be applied to the other. Without that, the stereo image would make sudden turns to one or another side.
<b>24. When mixing a popular music recording using an automated console. If the console is neither digital nor digitally controlled analogue, what facilities would you expect to be able to automate?</b>
Channel muting.
Anything else
<b>25. Often a multitrack recording contains unwanted sounds when instruments are not meant to be playing. State how this can be rectified.</b>
a) apply automated channel muting
b) use a noise gate
<b>26. On a VCA automated mixing console a number of level changes have been made to the vocal. The producer wishes to keep these changes but increase the overall vocal level by 3dB. Explain how this is achieved.</b>
a) Use fader grouping. Make the vocal channel fader a slave to another, group master fader. Use the master fader to increase the overall level.
b) If there is a compressor inserted in the vocal channel, increase its make-up gain by 3 dB.
<b>27. Is it possible to have an automation system which has moving faders yet the gain is altered by VCAs?</b>
Yes. The faders provide visual feedback in such a system.
<b>28. Describe the Decca tree system of miking.</b>
The Decca tree miking involves two side microphones positioned a few meters apart and one center microphone closer to the stage by about 1.5 meters. The center microphone is used to get a clearer center image than that given by a spaced stereo pair. Originally omnis, the microphones are now used of any polar pattern. With omnis it is desirable to aim the mikes slightly inward and downward, since omnidirectional microphones become more directional as the frequency increases.
<b>29. Describe why `update' or `trim' mode is desirable in a mixing console automation system.</b>
It is good for fine tuning the automation data. Update/trim mode allows one to literally add automation data on top of the existing data - the old data will not be erased or overwritten, just the changes added. It is like cumulative write, seems.
<b>30. State the meaning of EACH of the following sets of initials
i) VU ii) PPM. Briefly describe the difference between VU and PPM meters.</b>
VU = Volume Unit. It is an integrating (averaging) meter, aiming to correspond more or less to the perceived loudness of the signal. As an ear responds to the average rather than to the instant level, it takes 300 ms for the VU meter to establish its reading. So the instant peaks will pass unnoticed.
PPM = Peak Program Meter. PPM is a meter with much faster attack, it has more to do with the actual level than with the loudness.
<b>31. With reference to an analogue multi-track recorder briefly explain a) the term `sync' output b) why the `sync' output is required when overdubbing.</b>
Analogue tape recorders utilise separate heads for recording and playback. Although these have different characteristics, the operating principle is the same. So a signal can be played back through a recording head also, albeit with reduced HF response.
a) The term ‘sync’ output refers to playing back a track through the recording head.
b) If foldback signals were taken from the playback head during overdubbing, it would result in a significant delay between what was recorded and what was played back. This is due to the fact that it takes some time for the tape to travel from the recording to the playback head.
<b>32. Describe how a noise gate can be used to create gated reverb on a snare drum, using two microphones and the natural reverberation of the studio.</b>
Place one of the microphones so it catches the snare drum sound directly and the second more far to look for the reverberating sound. Bring the mikes to the mixing console, one channel for each. Let’s call them a snare channel and a reverb channel. Insert a noise gate into the reverb channel insert point. Use auxiliary send of the snare channel to send part of its signal to the external key input of the gate. Mix the two channels as appropriate. Now when there appears a signal at the snare channel (and AUX send), the gate opens and the natural reverb caught by the second microphone finds its way to the mix.
<b>33. Briefly explain, in terms of quality, the effect of mixing a signal with a version of the same signal delayed by 1 millisecond.</b>
This would result in a hollow sound due to phase cancellation. Phase cancellation or comb filtering means frequencies being added or subracted according to their phase relations.
<b>34. State the effect that would be achieved by continuously modulating the delay time up and down. In reference to question 33.</b>
Flanging.
<url>http://www.soundonsound.com/sos/jan01/articles/vintage.asp</url>
<b>35. Comment on the use of walking surfaces in audio drama production.</b>
Foley studios are equipped with different surfaces like sand, concrete, glass, gravel. These are used for producing sound effects (footsteps, movements etc) in accordance to what is going on at the screen.
<b>36. Why is it not suitable to connect the output of a record player pickup cartridge directly to a studio mixing console? Give two reasons.</b>
1) It has a low output level, 0.5 mV to 5 mV. This needs to be amplified, a line input typically uses 100 mV or so.
2) The LP mastering utilises a special equalisation called RIAA equalisation to overcome some physical limitations of LP recording. On playback, this should be reversed.
<b>37. If it is desired to equalise or compress an entire stereo mix, to which inputs/outputs of the console should the equaliser or compressor be connected? Assume a fully specified mixing console.</b>
To the stereo output (master) channel insert points.
<b>38. Describe the function of the stereo link switch in a compressor.</b>
Stereo link switch, when turned on, makes the two channels to have the same compressing behaviour. When levels on one channel induce compressing, the same action would be carried out at the other channel. This is good for avoiding sudden left/right turns of the stereo image.
<b>39. What is VCA Grouping?</b>
VCA grouping is a mechanism for allowing one, master fader, to control the level of other, slave faders. It differs from traditional channel subgrouping in that no audio signal passes through the master fader, ie the channels are not mixed to the master. The function of the master fader is to provide control voltage to the channel VCA-s.
<b>40. Explain flanging.</b>
Flanging is an effect where two identical signals are mixed together while being continuously and smootly shifted in phase. At times there is one signal slightly ahead of the other, then the opposite way. As a result, everchanging comb filtering occurs producing hollow and spaceous sound. Delay times are kept relatively short, up to 20 ms.
<b>41. Explain chorusing.</b>
<url>http://www.soundonsound.com/sos/jan98/articles/learnprocessors.htm</url>
Chorus is an effect where two signals of roughly the same level are added together, while one of them being pitch modulated by an LFO and slightly delayed. The idea behind chorusing is to pretend as if there were more players/singers performing the same part. The delay time is longer than in flanging, 15-35 ms, since the purpose here is to get the two signals nicely merge into one, not to produce comb litering. The side-effect of chorusing is de-localisation of the sound - it is not clear where the sound comes from, it seems to move further back in the mix.
<b>42. How can chorus be created with two tape machines.</b>
Play back the same material on the two machines, but start them at slightly different times. If the machines (or one of them) have varispeed control, play with these to induce subtle pitch changes. Mix the outputs from the machines together.
<b>43. Where does the term flanging come from?</b>
It comes from the technique used with analog tape machines to produced the flanging effect. Two machines were put to replay the same material and the delay between them was modulated by applying pressure to the supply reel flange. This was done alternatively on both machines.
<b>44. What is noise reduction?</b>
<url>http://www.soundonsound.com/sos/1996_articles/jan96/reducingnoise.html</url>
<url>http://www.soundonsound.com/sos/mar99/articles/20tips.htm</url>
On the context of analogue tape recording, noise reduction deals with reducing the subjective amount of tape background hiss during quiet passages. This is necessary to achieve greater usable dynamic range.
<b>46. List some milestones in noise reduction technology.</b>
<url>http://www.dolby.com/company/chronology1980_1989.html</url>
1965 - the first demonstration of Dolby A-type noise reduction system (to Decca Record Company).It uses four separate frequency bands and selective pre-emphasis - boost is applied only to frequency bands where the level falls below a specific threshold. to deal with. Used in professional systems only. Up to 15 dB noise reduction.
1966 first commercial recording with Dolby A (Vladimir Ashkenazy playing Mozart piano concertos).
1967-68 - Dolby B, consumer level system was developed. Once again, selective pre-emphasis but only one frequency region. The frequency above which the boost is applied depends on spectral content of the material. Noise reduction max 10 dB.
1970 - first pre-recorded cassettes with Dolby B system released.
1980 - Dolby C-type system introduced. It is more relevant to home recording than B. Effective down to 100 Hz, though most of work is done between 1-10 kHz. Anti-saturation circuitry included to prevent high frequency overload when boost is applied to signals already having high level of HF content. Up to 20 dB noise reduction.
1986 - Dolby SR (Spectral Recording) introduced. Up to 25 dB noise reduction. 10 filters used to divide the signal into bands, some of them fixed while others changing along with the spectral content of the material. The system always ensures maximum possible energy recording in each band. Anti-saturation mechanism is also incorporated. A high-cost system, generally found only in professional environments.
1989 - Dolby S demonstrated. Derived from SR (inherited some of the filter technology from it) and C type systems. Semi-pro system like Dolby C.
<b>47. Describe what is meant by plate reverb and spring reverb.</b>
Plate reverb unit consists of a thin metal plate located in the sound-proof enclosure. The plate is put to vibration by a transducer mounted on it. Two other microphone-like transducers catch the vibrations reflected from the plate edges. (Two are required for stereo.) The reverberation time is adjusted by varying the pressure of a damping pad against the plate - higher pressure causes the vibrations to cease more quickly.
Spring reverb operates similar to the plate reverb, but uses long metal spring for vibrations/reflections instead of plates. Once again, the initial vibrations are transferred to the spring by a loudspeaker-like transducer. Nowadays rarely used. Can have a great effect on strings, but sounds awful on percussive sounds because of unavoidable twanging.
<b>48. Explain the term "in-phase".</b>
Two signals are in-phase at a specific point when both of them are at the same part of their oscillating cycle at that point. More simply, when one oscillating signal has its maximum (minimum) at a specific time and location, and the other signal also has maximum (minimum) value at the same time and place, the two signals are in-phase. The opposite would be out-of -phase, when the maximum value of one signal meets the minimum value of another signal at some specific point. Their sum then would equal zero.
Cheers,
mommi
Last edited by mommi on Fri Oct 10, 2003 9:37 pm; edited 1 time in total |
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Wed Oct 08, 2003 10:02 pm |
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Albow
Joined: 02 Sep 2003
Posts: 27
Location: Spain |
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Seemed to come together a bit better this week, and I even seem to understand the odd concept regarding automation. Of course I may yet stand corrected 
1. What is the function of automation?
Automation enables changes to be made to the mix in realtime, with the
feature of being able to record the changes. Particularly used with
faders.
2. Why would faders have to be moved during the mix?
Because the sound levels of individual instruments would have to be uppered
and lowered during the song. This could be to give prominence to certain
instruments at certain points in the song. It also could concern the whole
sound, for example, the song might start off with a few instuments or a
single instrument, which is loud at the beginning to give the intro an
impact, and would be faded out more as other instruments are brought in to
the sound.
3. What aspects of mixing are normally automated on an analogue console?
Faders, muting, aux send, EQ.
4. Comment on VCA vs. Moving Fader automation.
With moving faders, the signal is linked directly to the fader knob/switch,
which can be touch sensitive and be manually moved, or will move by
themselves in automation. With VCA, the gain control amplifies or lowers a
voltage signal of the channel - no audio signal is passed through the fader
and the faders can not be seen to move. Performance with moving faders is
considered to be better, but with advances in VCA automation, this is now
contested.
5. Describe write/update/read.
These are the modes that automation can be set to. Write mode will take
any change that takes in automation in realtime, will implement it and
record that change Update mode will allow you to confirm the changes you
have made, or modify them further. Read mode will allow the user to see
the current automation configuration, but not make any changes, or change
back to previous settings.
6. How does an automated mix session start?
By approximating the fader settings for the performance and use it as a
standard snapshot to be returned to if needed.
8. How would a section of automation data be re-recorded?
By going into update mode, playing back the performance, and making the
necessary modifications. The new recording then can be compared to a
snapshot of the previous recording.
9. How are automation 'punch-ins' blended smoothly with existing data?
By using crossfades from the existing recording to the start of the punch
in, and back out from the end of the punch in.
10. How could a complex series of moves be increased in level by say 3dB?
By using fader grouping. Include the series of moves under one group and
then raise the group by 3db.
11. Describe fader grouping.
This is collecting several single fader channels under one fader group. It
could be done, for example with all the drum faders. If everyone is happy
with the drum sound as a whole, this could be put under one fader group and
the volume adhusted as required.
12. How could EQ, for instance, be automated using an analogue mixing
console?
Adjust EQ during playback in write-enabled mode.
14. How would EQ be automated using a digital mixing console?
On a digital mixing console you would select the type of EQ required, and
use write, read, update modes
15. What is 'recall'?
Recall is used to call back a previously saved snapshot of a track
16. Why is recall necessary?
It is necessary, so that we can see what changes have been made to a track
and decide if they have worked well or not.
17. What advantages does a digital console have over an analogue console
regarding recall?
With a digital console, one can take as many snapshots as required and
recall them all, giveing more storage to try doing a take in whatever way
required. They then can all be compared.
18. Describe how you would use a compressor with a side chain input as a
de-esser.
By taking a signal from a vocal channel insert to a compressor and also
routing the vocal signal through a filter and then to the compressor side
chain. This works by tuning the filter to only allow the offending
frequency to pass, when the compressor sees the signal on the side chain it
"ducks " the output from the compressor. Now you have a compressor that
only "ducks " the sibilant (hissing) content of the vocal.
19. Describe how you could use a two-channel compressor with a stereo link
switch as a de-esser without using the side chain inputs.
Put the normal signal through one channel and the higher frequencies
through the other channel, setting the threshold lower on that channel.
20. Explain why a noise gate would not be used when recording Scottish
bagpipes.
A noise gate is used to stop unwanted noise during quiet passages in the
music. The bagpipes have constant drones and sustained notes which are
essential components of the character of the bagpipes sound. You would not
want the noise gate to cut these off so it would not be useful.
21. What noise is often heard from an orchestral harp, besides the sound
of the strings?
The plucking sound of nails on strings, and the sound of feet tapping the
pedals on the floor
22. Describe how you would use a noise gate to impose a rhythm onto a
sustained synth string sound.
Similar to obtaining the choppy guitar sound, you could sent out the drum
sound signal through a side chain and feed it back into the gate, chopping
it up by means of a short envelope, activated when the drum sound is made.
23. Describe why the two channels of a stereo compressor must be linked
when processing a stereo signal.
Because the compression would have a different effect for each channel
individually, and this would lead to different levels on each side. With
the two channels linked, the sound is unified.
24. When mixing a popular music recording using an automated console. If
the console is neither digital nor digitally controlled analogue, what
facilities would you expect to be able to automate?
Faders, EQ, panning.
25. Often a multitrack recording contains unwanted sounds when instruments
are not meant to be playing. State how this can be rectified.
By using a noise gate.
26. On a VCA automated mixing console a number of level changes have been
made to the vocal. The producer wishes to keep these changes but increase
theoverall vocal level by 3dB. Explain how this is achieved.
By making the changes needed of the vocal performance, bouncing the whole
track and turning the volume up by 3db for the playback of the track. Then
record the track at that level.
27. Is it possible to have an automation system which has moving faders
yet the gain is altered by VCAs?
Yes. In this automation system, the faders will move by mimicking the
changes occurring in the VCAs, but the control is exclusively with the
VCAs.
28. Describe the Decca tree system of miking.
A triangle of microphones is placed from 10 to 12 feet above the stage
level just behind the conductor. There is often also 2 microphones
flanking the orchestra, or placed one third of the way in from the width
edges of the hall. In the mix, the centre mike is distributed equally to
both stereo channels, with the right and left branches going to their
respective stereo channels
29. Describe why `update' or `trim' mode is desirable in a mixing console
automation system.
It is desirable because one can make changes incrementally, making minor
modifications as opposed to having to start afresh with automation if not
totally happy.
30. State the meaning of EACH of the following sets of initials i) VU ii)
PPM. Briefly describe the difference between VU and PPM meters.
Volume Unit Peak Program Meters
Both display signal volume levels. The Volume Unit displays the average
signal whereas
31. With reference to an analogue multi-track recorder briefly explain a)
the term `sync' output b) why the `sync' output is required when
overdubbing.
The sync output ensures that the track that is now being laid onto an
existing track is in syncronisation with that track. In the analogue
multi-track recorder the record head and the playback head are separate
from each other, so there will be a delay incorporated into the recorded
version if sync is not activated.
32. Describe how a noise gate can be used to create gated reverb on a
snare drum, using two microphones and the natural reverberation of the
studio.
One mic would be placed on the snare, another a few feet away, perhaps
suspended from the ceiling to capture the room sound. You would then gate
the room mic.
33. Briefly explain, in terms of quality, the effect of mixing a signal
with a version of the same signal delayed by 1 millisecond.
The delay will not be picked up by the ear, but it will give phase problems
and a slight flanging effect, making the sound appear thin and hollow.
34. State the effect that would be achieved by continuously modulating the
delay time up and down. In reference to question 33.
This would make the sound wobble like a flanger was connected.
35. Comment on the use of walking surfaces in audio drama production.
Walking surfaces are provided by the Foley room. A number of sufaces are
either laid down or improvised (eg for the effect of walking on snow , rock
salt can be used). This is because to mike the steps in actual performance
is problematic. It is far easier to achieve the required sound in the
studio and work on the sound, than to try to mic-up somebody to record
their footsteps and/or the enviroment around them which could produce any
number of sonic imponderables.
36. Why is it not suitable to connect the output of a record player pickup
cartridge directly to a studio mixing console? Give two reasons.
The signal from the cartridge is too low to record - <0.5mV for a moving
coil and <5mV for a moving magnet. The signal is RIAA-industry
standardised to play vinyl records - this must be reversed to provide a
flat frequency response for line level output - you would then have to
boost the signal to 100mv to drive an amp.
37. If it is desired to equalise or compress an entire stereo mix, to
which inputs/outputs of the console should the equaliser or compressor be
connected? Assume a fully specified mixing console.
The console stereo bus' send and return.
38. Describe the function of the stereo link switch in a compressor.
The stereo link switch must be activated to make sure that the compression
in each channel is synchronised. If not, then the compression levels on
each channel will function independently of one another, and the sound will
not be unified.
39. What is VCA Grouping?
As with fader grouping, the VCA faders can be grouped together so that one
VCA channel is chosen to control the fading of all the VCA channels in that
group simultaneously as a group.
40. Explain flanging.
The sound of flanging is to add a spookiness to a note, giving the sound of
for example, a guitar line, a hollowness and a quiver. It is done by
adding a delay of 0-20ms.
41. Exlain chorusing.
The effect of chorusing can be to fatten out a note. It is often used to
make strumming a 6-string guitar sound like a 12-string, as the notes
played ring out more. It is done by delaying the sound by between 15 and
40ms.
42. How can chorus be created with two tape machines.
Connect the 2 machines and send the signal from one to another, starting
playback at different times.
43. Where does the term flanging come from?
When the practice was originally used, it involved applying pressure to the
flange of the supply wheel to produce the delay.
44. What is noise reduction?
The reduction of background noise and sounds extraneous to that being
played on the recorded medium. An example would be the accompanying hiss
of a tape contact with the tape head. When the noise is reduced the
possibilties for reproducing dynamic range are greater.
46. List some milestones in noise reduction technology.
Dolby pioneered the term, by compressing the recorded sound and reversing
it on playback. Throughout the mid sixties it went from being added to
prefessional systems to home recording tape systems. This was Dolby B and
was available on most home stereo systems. Dolby then introduced surround
sound first used for motion picture soundtracks. IN 1977 Dolby SR was a
step further for incorporating a larger dynamic range in sounds. This was
achieved through the CP50 sound processor which gave the then superior
soundtracks of Star Wars and Close Encounters.
1992 Dolby Digital (AC-3) provided a multichannel digital audio coding
technology first used for cinema sound (1992). Today it is also used to
bring multichannel sound into the home via a wide variety of digital
formats, including DVD, DTV, digital cable, and DBS. Today Dolby Digital
technology makes it possible to place a multichannel digital soundtrack on
35 mm release prints also brings multichannel sound into the home via
consumer digital formats.
47. Describe what is meant by plate reverb and spring reverb.
Plate reverb is produced by placing a small speaker on a metal plate and
micing the output. Spring reverb is produced by running an electric
current through a coil.
48. Explain the term "in-phase".
Two signals are in phase when their oscillating cycles are in tandem. They
therefore achieve high and low points in their cycled at the same point in
time. In practice this gives us a clear unified sound. If the signals
were out of phase then the sound would not be as clear and would appear
distant. |
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Thu Oct 09, 2003 10:32 pm |
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julesf
Joined: 31 Aug 2003
Posts: 58
Location: Southampton, UK |
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Hello Happy Bunnies 
RT Week 6
<b>1. What is the function of automation?</b>
To allow aspects of the mix to altered in real time, the parameters that are changed
are memorised so that the final mixdown is completely automated. Basic automation includes
Fader movement and mutes, however a console with quality automation will include Eq, aux sends, pan controls,
And many other parameters to be altered and memorised. These setting are stored and can be recalled later (snap shot). In the early days engineers used to photograph the console between projects to enable the console to be reset to the projects settings when the next session was due. This was a time consuming process. Automation avoids the need for these measures. The project settings can be save onto magnetic media and instantly recalled at anytime.
<b>2. Why would faders have to be moved during the mix?</b>
It is normal for many parameters to change during the mix. This may be a simple level change, or aux setting increased for effect, a pan from L-R or R-L for example. During mixdown the engineer and producer work together to provide smooth fades and other adjustments. Before automation all of these parameters had to be changed manually with many hands on the console. Automation has revolutionised the mixdown process so as one engineer can carry out all of the complex adjustments to the mix.
<b>3. What aspects of mixing are normally automated on an analogue console?</b>
Certainly volumes and mutes. But on a good console more parameters are often automated.</b>
<b>4. Comment on VCA vs. Moving Fader automation. </b>
Voltage controlled amplifier automation gives some level of control, however moving
Fader automation despite being the more expensive approach, actually is accepted as the more natural to
Way to work with automation. Here the fader will always "fly" to its correct setting; this provides a visual feedback to the fader position. VCA faders stay put and don't move although the level for that fader's channel will reflect the actual saved automated setting. This can be very confusing. Often LCD displays are used to show a graphic reference of actual fader position because the faders cant.
<b<5. Describe write/update/read. </b>
Write sends the movement information of any automated process to storage.
Update allows the modification of any previous write activity to be altered or amended.
Any movement of additional parameters that have not been previously stored are ignored.
<b>6. How does an automated mix session start?</b>
Automation may start with the recall of a standard snapshot. Also time code to the recorder
or sequencer must be established. The mixdown is then analysed section by section to establish what should be changed and where. Ideas can then be tried and automated.
<b>8. How would a section of automation data be re-recorded?</b>
By using the update facility. Update would be selected and at the point where automation
Was recorded the changes could be made.
<b>9. How are automation 'punch-ins' blended smoothly with existing data?</b>
by the use of crossfades or by selecting a zero point crossing.
<b>10. How could a complex series of moves be increased in level by say 3dB?</b>
By assigning the relevant channels to sub groups and then raising the subgroup fader by an amount
Equivalent to a 3db increase.
<b>11. Describe fader grouping. </b>
Fader grouping allows a number of channels to be subgrouped to one fader. This can be very useful for say, drums.
A number of channels are used for drums respectively. If you want to increase the snare in the mix then this is easy, just raise the snare fader. However if you wish to increase the whole drum set in the mix but keep the actual drum mix between drums the same, then it is a good idea to assign all drum channels to a drum sub group. Now if you wish to increase the drums in the mix simply raise the drum subgroup fader.
<b>12. How could EQ, for instance, be automated using an analogue mixing console? </b>
Firstly the console would need to be of an automated design and be synced to the recorder by "SMPTE"
MIDI clock or WORD clock. The automation would need to have reference to real time so as the automation could be synced to the audio. Then the Eq could be altered in real time by using the Write, read, and update buttons and turning the control knobs on the console.
<b>14. How would EQ be automated using a digital mixing console?</b>
The process is very similar however often digital desks do not have controls for all of the functions on the desk. It is necessary to click through menus and find the appropriate Eq which is then altered using a set of controls that are common to all processes or by dragging the settings virtually. However desks such as the Mackie D8b have a layout and feel very similar to an analogue desk but with all of the functions and features (including flying faders) of a digital desk.
<b>15. What is 'recall'?</b>
Recall allows a saved "snapshot" of stored settings to be recalled.
<b>16. Why is recall necessary?</b>
Without recall you would not be able to retrieve previously stored settings.
<b>17. What advantages does a digital console have over an analogue console regarding recall? </b>
More parameters such as digital FX plugins can be stored and then recalled for the project.
the recalled settings will include plugin processors such as compressors, reverbs and a whole array of other features. On the analogue desk, just the control knob settings can be recalled.
<b>18. Describe how you would use a compressor with a side chain input as a de-esser.</b>
<b>19. Describe how you could use a two-channel compressor with a stereo link switch as a de-esser without using the side chain inputs. </b>
The right channel is fed a signal from the vocal channel via a suitable filter tuned to the sibilance. The second channel just treats the vocal channel on an insert. The channels are linked using the link switch. Here the filtered channel 1, acts as the control where channel to is the controlled. The compressor will only "duck" the vocal on the controlled channel two when the sibilance passes the filter.
<b>20. Explain why a noise gate would not be used when recording Scottish bagpipes. </b>
A noise gate is designed to shut off the signal path when the amplitude of a signal drops below the pre-set threshold.
This will prevent the unwanted noise floor being heard in pauses say between vocal lines. The Scottish bagpipe has a droning tone, which sounds all of the time and is part of this instrument character. Therefore there is simply nothing to gate off.
<b>21. What noise is often heard from an orchestral harp, besides the sound of the strings? </b>
The sound of the pedals can sometimes be heard when recording Harp. Careful use of mic-ing techniques can help to avoid the this unwanted sound being recorded, though some may say that it is produced naturally along with the instrument and therefore is not a problem.
<b>22. Describe how you would use a noise gate to impose a rhythm onto a sustained synth string sound. </b>
Patch the synth through the noise gate. Send the output of an LFO or the output of a drum machine into the side chain input of the noise gate. Adjust all parameter variables to taste.
<b>23. Describe why the two channels of a stereo compressor must be linked when processing a stereo signal.</b>
By activating the link switch and controlling channel B with the VCA control from Channel A allows both channels
Of a stereo mix to be compressed equally which will sound more uniform.
<b>24. When mixing a popular music recording using an automated console. If the console is neither digital nor digitally controlled analogue, what facilities would you expect to be able to automate? </b>
Faders, mutes and may be aux sends.
<b>25. Often a multitrack recording contains unwanted sounds when instruments are not meant to be playing. State how this can be rectified. </b>
If using a DAW then unwanted sounds can be edited out in the arrange window. If the recording is Analogue then
It may be possible to erase the unwanted parts. Automated mutes can be programmed. Otherwise use a noise gate on the tracks.
<b>26. On a VCA automated mixing console a number of level changes have been made to the vocal. The producer wishes to keep these changes but increase the overall vocal level by 3dB. Explain how this is achieved. </b>
Assign the relevant channel(s) to a subgroup fader and raise this fader to allow a 3-dB gain to the overall mix for that subgroup.
<b>27. Is it possible to have an automation system, which has moving faders yet the gain is altered by VCA's? </b>
Electronically yes, the faders could be stepped in the conventional way but the resistance change from the fader could send a varying signal to an array of VCA's which in turn would alter the levels. Whether this system is used I am not sure.
<b>28. Describe the Decca tree system of miking.</b>
Three Omni Microphones are set up in a triangular layout and placed behind the conductor and about 10-12 feet above the stage. The two outside mic's are angled down at around 30 degrees pointing at the stage and are panned left and right in the mix respectively. The centre mic is panned to centre. This system was developed by Decca in the mid / late 50's and is still used today. Often outer "Flanking mic's" and spot mic's are added to the mix for reinforcement.
Hear is another excellent link: [url] http://www.ransom-recording.co.uk/recording%20techniques2.htm[/url]
<b>29. Describe why `update' or `trim' mode is desirable in a mixing console automation system. </b>
Update or "trim" allows previously recorded automation data to be finely tuned. It may be that a producer is trying to obtain a repeat delay on part of a vocal and the first automated take was not quite correct. Update allows for the automation to be slightly corrected without having to completely redo the procedure.
<b>30. State the meaning of EACH of the following sets of initials </b>
<b>i) VU ii) PPM. Briefly describe the difference between VU and PPM meters. </b>
VU meters measure the average level of the signal; PPM meters read the peak signal level
Another julesf classic find, Amek!
http://www.amek.com/oldsite/datashee/meters.htm
<b>31. With reference to an analogue multi-track recorder briefly explain a) the term `sync' output b) why the `sync' output is required when overdubbing.</b>
A professional analogue recorder uses three heads, Erase, Record and Replay. These heads are all in a line, as the tape passes through the track for overdub is erased and then it passes over the record head where the new audio for the take is recorded. Then the tape passes over the replay head whereas the recorded sound can be heard. Because of the gaps between the heads and the fact that the tape is moving (normally at 15 or 30 IPS) there is a delay between the sound being recorded and the sound being replayed. This will cause the musician to play an overdub later than it should be and when played back the track will be out of sync. To overcome this limitation the sync facility is switched in while overdubbing. This effectively uses the record head to play back all of the other recorded tracks while the new track is recorded. Because all of the monitoring and recording is being monitored using the same head no delays are created. Because the record head is optimised for recording the playback quality is not so good as using the record head. This is irrelevant as it is only being used for monitoring. When mixing down the sync facility can be switched back to playback mode whereas full sonic quality will be restored for mixdown.
<b>32. Describe how a noise gate can be used to create gated reverb on a snare drum, using two microphones and the natural reverberation of the studio.</b>
The snare drum is miked conventionally into a recording console channel. A second mic is placed to pick up studio live room reverb ambience from the kit and this is sent through a noise gate to a second channel or an aux return.
The side chain on the gate is sent a signal from the snare drum mic. When the snare drum is hit hard enough to produce a signal to trigger the noise gate then the gate will open and the studio ambience will be also mixed with the dry snare sound.
Another way of doing this to obtain a cleaner snare reverb is to send the snare mic channel via a post fade aux control to an amp and speaker placed in the area of the live room or an echo chamber. The gated snare ambience mic will now react to the reverb ambience sound coming back from the snare sound delivered by the speaker. The noise gate can thus in this case control from the ambient mic input as opposed to the side chain signal from the Snare mic.
<b>33. Briefly explain, in terms of quality, the effect of mixing a signal with a version of the same signal delayed by 1 millisecond. </b>
The sound will fatten and go a little muddy due to comb filtering.
<b>34. State the effect that would be achieved by continuously modulating the delay time up and down. In reference to question 33.</b>
A phasing effect would be heard, this sometimes used to be referred to as "jet planing"
35. <b>Comment on the use of walking surfaces in audio drama production. </b>
A walking surface is used to produce a sound that is realistic to the surface of the scene. In some cases it is
Not possible to have a surface in a production studio that will produce the correct sound for the scene. So an imitation walking surface is used which is loaded with the correct media to produce the sound for the scene. The walking surface is then operated by a "Foley" artist and the sound "dubbed" on to the sound track of the film.
<b>36. Why is it not suitable to connect the output of a record player pickup cartridge directly to a studio mixing console? Give two reasons.</b> the output of a record deck cartridge is very small typically 2 - 3 mv. Most mixer pre-amps are not optimise to amplify this smaller signal and turning up the gain trim very high may mean unacceptable noise floor. Also due to the mechanical limitations of the phonograph reproducer a special form of Eq is added known as RIAA. The pre-amp used should be of the RIAA type to achieve faithful reproduction and quality.
<b>37. If it is desired to equalise or compress an entire stereo mix, to which inputs/outputs of the console should the equaliser or compressor be connected? Assume a fully specified mixing console. </b>
Left / Right main mix out to the inputs of the effects unit(s), output of the effects units to the input of the mix down master machine. Unless the console is equipped with Left / Right mix inserts (not seen those before) in which case these can be used.
<b<38. Describe the function of the stereo link switch in a compressor. </b>
The stereo link switch allows channel A VCA control voltage to control Channel B VCA as well.
So as the compressor works as a stereo pair.
<b>39. What is VCA Grouping? </b>
VCA Grouping is the term used when the control signal from one VCA is linked to control others. In this case all other VCA's linked will follow the characteristic of the master signal and control their levels in the same way.
<b>40. Explain flanging.</b>
Flanging is an effect caused by phase shift and feedback. The effect was produced by setting up two tape machines which both had a copy of the material to be "Flanged". The two machines where played back in time when the flanging was required the engineer simply placed his or her finger on the flange of the supply reel and slowed down one machine and then let it speed up again. This put the sound out of phase also some of the phased signal was fed back to the original signal which enhanced the flange effect. Of course all of this is now done electronically or digitally.
<b>41. Explain chorusing. </b>
Chorus is produced by using two identical source sounds, and introducing a delay of about 25 - 30 ms to one sound source.
An LFO then sweeps the pitch slightly and the delayed sound is then mixed back with the original. The sound is supposed to produce a kind of ensemble effect for vocal but it is generally best used on guitar. It can be used slightly to thicken a vocal but the side effects cause the sound to become muddy and a bit distant.
42. How can chorus be created with two tape machines.
Use two tape machines playing the same source sound, delay one by using the pitch control or slowing the tape with a finger.
<b>43. Where does the term flanging come from? </b>
By the placing of a finger on the "flange" of the tape reel.
<b>44. What is noise reduction? </b>
Noise is an inherent part of the process of amplification and magnetic recording.
Amplification noise is generated from electronic components and the electrons themselves.
Magnetic tape noise is produced by the magnetic flux on the tape. Most is actually "White noise"
Which is a mixture of many frequencies and can be heard as a hiss. Higher quality circuits produce less
Noise and in most modern circuits the signal to noise ratio is such that it causes no issue. Noise was a problem and many filters have been produced to combat it. Noise gate circuitry can be used to disguise noise floor levels.
In analogue tape where noise can be a problem filters and pre-emphasis where used. But the most commonly known company researching solutions for tape noise was "Dolby Laboratories" Dolby made a number of different noise reduction solutions for tape based systems known as Dolby A, Dolby B, Dolby C, Dolby S, Dolby SR. The first systems were based on "Selective Pre-emphasis" which boosts high frequencies during recording and then cuts them during playback. This was how Dolby A worked; it was used for commercial systems only. The later B and C systems were based on compander circuitry which compressed the higher frequency noise floor during quiet passages but did not destroy The frequency range at higher listening levels. The SR system was very complex and attempts to treat the whole frequency spectrum rather than selective parts of it as the others do. The systems generally were capable of reducing noise by 10db for the early systems right up to about 25db. Tascam also produced a system known as DBX which involved 2:1 compression and noise reduction for their semi-pro tape machines. This system reduced noises by up to 30db but it was only used for these recorders due to compatibility issues between tape and alignment.
<b>46. List some milestones in noise reduction technology.</b>
Most systems that were used over the years just involved high frequency filtering,
However Dolby revolutionised noise reduction:
1965 Dolby A demonstrated to Decca.
1966 the first commercial release using Dolby A.
1968 First consumer product fitted with Dolby B, A KLH Tape machine.
1969 development of the compact cassette using Dolby B.
1970 first pre recorded compact cassette released by Decca and Sony using Dolby B
1971 Clockwork Orange, first film released using Dolby noise reduction.
1974 35mm Dolby optical soundtrack introduced at the SMPTE Convention.
1974 Dolby used for FM broadcasting.
1975 first Stereo optical Dolby film soundtrack release.
1979 Dolby receives first award.
1979 HX headroom extension for cassettes introduced.
1980 Dolby C introduced
1981 over 1 Million Dolby C processors fitted into products.
1981 Dolby SP introduced.
1989 Ray Dolby awarded Oscar then EMMY award.
1989 Dolby consumer S type demonstrated.
1990 First 1" 24 Track with Dolby S
1995 First Radio production in Dolby surround
http://www.dolby.com/company/chronology1965_1969.html
<b>47. Describe what is meant by plate reverb and spring reverb.</b>
Plate reverb utilises a suspended steel plate fitted with transducers. Sound is input to the transducers
caused the plate to vibrate, these vibrations are collected by the transducers at the other end of the plate.
A delay is produced in the process that makes a nice vocal reverb.
A spring-line unit utilises a supported spring with a transducer at each end. The transducer drives sound waves into the spring, which are picked up by the transducers at the other end of the spring. The spring causes a delay, which is proportional to the springs, length. This when mixed back with the original sound produces an echo. This is good for, and is still used today on many guitar amplifier combos.
<b>48. Explain the term "in-phase". </b>
When two identical waveforms are amplified and line up accurately on the positive and negative parts of the cycle they are said to be "in phase" if a delay is added to one of the signals then it with start to go out of phase. This will lead to the signals producing phase shift characteristics, which will change according to the phase shift. The sound, of phasing was sometimes referred to as "Jet-Planing" as it sound like the phase shift that can be heard from a jet engine on take off and is being produced by the effects of comb filtering. When the signals are exactly 180 degrees out of phase, phase cancellation will occur which will cancel out and reduce the signal level depending on the amplitude of the waves. If amplitude of the waves is equal then the sound will cancel completely. Quality Phasing link:
http://www.songstuff.com/Features/Technical/Phase/phase.htm/
See Yol Sunday  |
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Fri Oct 10, 2003 8:02 pm |
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AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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Very good work guys, I'm impressed by the constistancy you are delivering.
Regarding analogue automation, lets take that forward to the Live Workshop this week and have a chat about it there, it may be easier to grasp some of the concepts. I'm talking about using MIDI as the controller having been converted from SMPTE. The MIDI mute automation can be a very creative tool.
See you sunday.
Chris
_________________
It's all in the ears. - Learn the concepts not the software.
Audio Courses is a way into the music business for you
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Sat Oct 11, 2003 6:03 pm |
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Fenderocks
Joined: 09 Aug 2003
Posts: 26
Location: Music City, USA |
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1. What is the function of automation? It allows changes to the mix during the mix down, normally used with faders.
2. Why would faders have to be moved during the mix? To make one track quiet or louder, maybe to bring the beginning of the song to fade in after the intro. There lots of reasons to use faders during the mix either to hear different tracks louder then they would be in the final mix.
3. What aspects of mixing are normally automated on an analogue console? Eq, muting, faders, and aux send
4. Comment on VCA vs. Moving Fader automation. On the voltage controlled amplifier there are no audio signals passing through the fader automation. The moving fader automation will move the faders during automated mix down, therefore the signal moves through the fader.
5. Describe write/update/read.
Write- recording the automation information
Update- is used when you feel you need to make adjustments.
Read- after the final mix-down this comes into play once you listen to the tracks.
6. How does an automated mix session start? To set the faders for the performance and use snap shots to be returned to if needed.
8. How would a section of automation data be re-recorded? You make the necessary adjustments in update mode to compare to what was already recorded.
9. How are automation 'punch-ins' blended smoothly with existing data? Cross faders will help blend the punch-in together.
10. How could a complex series of moves be increased in level by say 3dB? I highlight all the information that needs to be louder and then go to the audio and then to 3db louder. But if you don’t have the option of that you can group the tracks to one fader, and then raise the group.
11. Describe fader grouping. Collecting single fader and grouping them together, if every fader that goes into the group everyone is happy with.
12. How could EQ, for instance, be automated using an analogue mixing console? Use two channels for one signal, then apply different eq settings. Then this would allow you to switch for either channel by automated fader or muting will result to different eq settings.
14. How would EQ be automated using a digital mixing console? In a digital mixing console you could highlight what you wanted eq to be added to and go to the eq settings.
15. What is 'recall'? Reading the settings that have been previously is stored.
16. Why is recall necessary? If you messed up, also if you what something to compare to.
17. What advantages does a digital console have over an analogue console regarding recall? A lot easier to apply different setting for one, also you can always bring up what you previously did up to as many takes as you want.
18. Describe how you would use a compressor with a side chain input as a de-esser. By taking a signal through a filter and then trough the compression side chain. Make the filter only to allow certain frequencies through, so when the compressor see the signal on the side chain it makes way for the output from the compressor.
19. Describe how you could use a two-channel compressor with a stereo link switch as a de-esser without using the side chain inputs. Few different ways, one by putting a signal through one channel and then throw the higher frequencies through the other channel putting the volume lower then the other channel.
20. Explain why a noise gate would not be used when recording Scottish bagpipes. By using noise gates on a compressor you would be taking away from the sound of the bag pipes by taking away the sustained notes and the drone of the bag pipes.
21. What noise is often heard from an orchestral harp, besides the sound of the strings? The sound of finger nails hitting the strings and the sound of pedal use.
22. Describe how you would use a noise gate to impose a rhythm onto a sustained synth string sound. By inserting the noise gate on the impose of a sustained synth string, this will allow the smith sound to go through until the output of the synth gets quiet.
23. Describe why the two channels of a stereo compressor must be linked when processing a stereo signal. So that both channels have the same amount of compression linked together in sync.
24. When mixing a popular music recording using an automated console. If the console is neither digital nor digitally controlled analogue, what facilities would you expect to be able to automate? Faders, eq, panning, as well as channel muting.
25. Often a multitrack recording contains unwanted sounds when instruments are not meant to be playing. State how this can be rectified. Use automating channel muting and noise gate, you also can clip by grabbing the edge of the track and bring it right up to the first note hit.
26. On a VCA automated mixing console a number of level changes have been made to the vocal. The producer wishes to keep these changes but increase theoverall vocal level by 3dB. Explain how this is achieved. Fader grouping, and you also can use compression to increase by 3db as well.
27. Is it possible to have an automation system which has moving faders yet the gain is altered by VCAs? Yes, and how they do this is by setting up the faders to mimic the changes in the controlled voltage amplifier. But still using the control by the VCA.
28. Describe the Decca tree system of miking. A triangle of microphones is placed from 10 to 12 feet above the stage level just behind the conductor. There is often 2 microphones
flanking the orchestra, or placed one third of the way in from the width edges of the hall. In the mix, the centre mike is distributed equally to both stereo channels, with the right to the right channel and left to the left channel.
29. Describe why `update' or `trim' mode is desirable in a mixing console automation system. Well if not satisfied with your finished part, you can easily bring up it again without having to start all over.
30. State the meaning of EACH of the following sets of initials
i) VU ii) PPM. Briefly describe the difference between VU and PPM meters.
Volume Unit and Peak Program Meters. They both display signal volume levels and with the Volume unit it show where the signal is.
31. With reference to an analogue multi-track recorder briefly explain a) the term `sync' output b) why the `sync' output is required when overdubbing. Sync output is, it make sure that what you are recording is in sync with the other track. The reason it is required is to obtain a sound that’s in time and sounds good.
32. Describe how a noise gate can be used to create gated reverb on a snare drum, using two microphones and the natural reverberation of the studio. One on the snare and the other hanging above the room two to three feet away, then you would put a noise gate on the suspended mic to allow the reverb to be captured.
33. Briefly explain, in terms of quality, the effect of mixing a signal with a version of the same signal delayed by 1 millisecond. It would be to fast to hear the delay effect however you would end up with a slight flanger effect or a hollowness sound because of the phase cancellation which means that frequencies would be added.
34. State the effect that would be achieved by continuously modulating the delay time up and down. In reference to question 33. It would be a flanger effect.
35. Comment on the use of walking surfaces in audio drama production. Walking surfaces are used to make sound effects towards movies, radio, or films. When you see movies and you hear the sound effect of someone getting there arm brake or even someone walking down the road that would be a use of a walking surface.
36. Why is it not suitable to connect the output of a record player pickup cartridge directly to a studio mixing console? Give two reasons.
1. The signal would be to low for recording
2. RIAA industry is a special equlazion to overcome physical limitations of lp recording and would have to be reversed.
37. If it is desired to equalise or compress an entire stereo mix, to which inputs/outputs of the console should the equaliser or compressor be connected? Assume a fully specified mixing console. The master channel, insert point of the stereo output.
38. Describe the function of the stereo link switch in a compressor. With a stereo link it will make the channels have the same compression. So when the compression is added it will mimic the same on the other channel.
39. What is VCA Grouping? Is where one master fader has the control over the other slave faders. The thing with though there is no audio signal is going through the master fader. The use of these is to control voltage with the VCA’s
40. Explain flanging. This happens when two identical channels are added together with one slightly ahead of the other, but not one of the channels is more the 20 ms ahead.
41. Exlain chorusing. Two signals around the same time added together. Then you take one of the signals and add a little delay and alter the pitch a little. Different from flanger with the delay time set at 15 to 35 sec.
42. How can chorus be created with two tape machines? Take your two tape machines play them both at the same time one a little before. If you can use varies tape speed to emulate the pitch change.
43. Where does the term flanging come from? Comes from the supply wheel to produce the delay by apply pressure to the flanger.
44. What is noise reduction? Deals with reducing any other noises, such as hissing during quiet parts or even on loud parts. Use to get a greater dynamic range too.
46. List some milestones in noise reduction technology. In 1977 Dolby SR a
step further for brining a larger dynamic range in all sounds. This was
achieved through the CP50 sound processor which gave the then amazing
soundtracks for Star Wars and Close Encounters.
In 1980 the Dolby C-type system brought in. It is more relevant to home recording than even a way to effectively bring down to 100 Hz, though most of the work is done between 1-10 kHz. The Anti-saturation circuitry was included to prevent high frequency overload when the boost was applied to signals already having high level of HF content, even up to 20 dB noise reductions.
47. Describe what is meant by plate reverb and spring reverb. To start it of use a thin metal plate in a sound proof enclosure. You vibrate the plate by what its mount on. The microphone captures the vibration from the plate. The spring reverb is made kind of like the plate reverb but by springs. The spring is picked up through using a loudspeaker like a transducer. It’s not used that much but that’s how you do it.
48. Explain the term "in-phase". Two signals are in phase when their oscillating cycles are in tandem. They then achieve high and low points in their cycled at the same point in
time. If the signals were out of phase then the sound would not be clear and would appear
distant. |
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Sun Oct 12, 2003 5:14 am |
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