Setting sound levels, theatre and live band mixing
Wed, 22 Apr 2009, 09:35 pmmike raine33 posts in thread
Setting sound levels, theatre and live band mixing
Wed, 22 Apr 2009, 09:35 pmBeing relatively new here, I was having a ramble around assorted forums when I came across a post that dwelt on the topic of sound levels in theatre. I noticed that it was locked, so I was unable to add my thoughts there, which, instead, I will do here. If this is a breach of this forum's etiquette, someone might let me know.
The start of that other discussion was this summary:
"In summary:
1. Your cd player will have a nominal line output, it may be fixed at 774mV as it is not usually a 600 ohm feed, but a 10Kohm feed (domestic).
2. Set the slide fader to 0dB which is the black line at the 3/4 mark (3/4 of 1 volt is 775 mV)
3. Set the main fader to 0dB which is the black line at the 3/4 mark (3/4 of 1 volt is 775 mV)
4. Adjust the channel gain which is usually a rotary knob so you get 0VU on the desk output meter.
5. Adjust you power amp input level, usually a rotary know, for the desired sound level in the theatre.
You now have the ideal compromise between best signal to noise (getting rid of the hiss) and not running out of headroom (distorting the sound coming out of the speakers."
I had a read of the link that the post also referred to.
The basis of my remarks stems from a couple of decades of live and studio sound mixing, so I have reasonable experience and expertise in the area.
My thoughts are these:
1 The principle behind the summary above and the Yamaha article is sound; i.e. optimise the gain structure from the source as much as possible. The aim of this principle is to maximise the signal to noise ratio. All elements in a signal path generate some noise, the effect is cumulative, and a maxxed-out power amp at the end of the chain will reveal all this clutter magnificiently.
2 However, there are some difficulties in the practical application of this principle, the main one being access to a power amp's level controls, if it has them (not all do). If the power amp is co-located with mixing desk, there is no problem of access. There is another problem, though. Long speaker cables from back of the hall (or thereabouts) to front of house speakers result in considerable energy and signal loss. The result is that the power amp may have to be driven harder to make up for this loss, therefore increasing the understory of noise. It is no accident that on music stage rigs, the amps are co-located with the speakers, not the desk, to minimise high-energy cable lengths.
3 Furthermore, over the last decade we have seen a huge increase in the use of active speakers, where the power amp is built into the speaker cabinet (which, unsurprisingly, is where the level controls are located).
4 In the past, power amps were bulky, heavy and noisy. They make great boat anchors. Current technology has resulted in smaller, lighter and virtually noise-free power amps (and active speakers). The importance of managing a venue's sound levels by manipulating the amp's controls is not as critical as it may have been once.
5 Within the fraternity of sound engineers with whom I'm acquainted, the common practice is to set the power amp levels to maximum, and to control overall level from the mixing desk's master control (which hovers around 0db). All audio peripherals (e.g. graphic EQs, compressors and so on) are set to unity gain (0 db), and are left alone.
6 Though this is a common practice, it is not universal. There are times when you know that you don't need the amp's full power. For example, I was mixing with my desk plugged into an in-house system that had so much power behind it that I would have been trying to mix with the bottom couple of millimetres of fader movement, which is just plain silly. So I set the power amp levels at a quarter, which was all that was needed for that particular job. There are also some rigs that are past their prime and very noisy . . . and again you have to balance between getting an adequate level without introducing too much noise.
7 I note that the Yamaha article talked about setting signal levels as high as possible to maximise signal to noise, but with the warning to avoid tape saturation or digital distortion. This notion has been in vogue for many years, but is not nearly as relevant these days. Current technology has reduced the intrinsic noise significantly, tape is rarely used, and 24 bit digital recording has increased the inherent dynamic range such that it doesn't really matter how low the signal is. The aim now is not so much to reduce noise, but to make sure that there is plenty of headroom for mixing.
8 If I were to recommend a general practice for setting levels, it would be this:
a) Set power amp to max
b) Set master fader to unity.
c) Set a desk input channel's trim to deliver the best signal without peaking
d) Bring the channel fader up to the desired level.
e) Only if the channel fader can't give a reasonable working range (i.e. it's too loud when it's nearly at the bottom) would I adjust power amp levels.
9 However, each venue is different, as is each sound system, so any practice has to be tailored to the combination.
Finally, one of my greatest thrills is to set up a rig, wind up the amps, return to the desk and listen to the absolute quiet. Then I put on a CD or something . . . and there it is, clean , crisp, undistorted and immensely satisfying.
Here is a quote from the
Sat, 25 Apr 2009, 09:15 pmHere is a quote from the above piece, you can follow the reasoning from the above if you are technically inclined, but it does prove my contention without any doubt,
"BACKGROUND YOU NEED TO KNOW
Realistically, audio signals at or near the noise floor of a system are not useful because the signal will not be
significantly louder than the noise. Therefore, some minimum usable level must be assumed below which the
electronic noise is considered objectionable. A signal to noise ratio of 20 dB is considered minimally acceptable
for good intelligibility. For a high quality system 30 dB would be a better figure to use. Using this value, the
range from this minimum signal level (30 dB above the noise floor) to the clipping level is the usable signal range
window for the system (also called the dynamic range in my way of thinking). However, for purposes of this
paper, the maximum output to noise floor is used as the dynamic range.
Every audio system with more than one electronic component has a "system gain structure". Gain structuring for
a system occurs in the signal processing chain between the mixer or another signal source and the power
amplifiers. One usual scenario is to set all the signal processors to unity gain and turn the amplifier inputs to
maximum. Unfortunately as you will see, given the different maximum outputs and noise levels of typical signal
processors, this method will may not come close to the best gain structure."
The article is an extremely good read, not understood by a large majority of sound operators.
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