Hi All.
This is sort of a summary/continuation of several questions I've asked in this room in the past. Thanks very much to all the AVIXA folks who have commented on these subjects in the past and brought me far enough to post all this. I hope it encourages healthy discussion. Hopefully I can find the clarity I've been looking for... and if my questions are apt, maybe the results will find their way into a future edition or addendum of the exam guide...
When specifying amplifiers I am constantly under pressure to reduce the wattage for cost reasons. I have struggled to understand several things about the 'Audio Principles of Design' chapter in the Exam guide as they apply to specifying amplifiers with a minimum of over-engineering.
At the same time, I have been part of system configuration in the field and found that my calculated taps are (at times) only barely enough to provide the 68dB-72dB my calculations accounted for. I would like to avoid under-engineering even more so than over-engineering.
There are four points of confusion for me:
1) Choosing appropriate headroom in the EPR calculation (p155)
2) Applying the RMS vs peak values yielded by the EPR calculation to amplifier selection (pp155-156)
3) Lack of clarity on the reasons we increase the summed tap wattages by 150% before selecting a 70V amplifier as shown in step #7 (p158, top).
4) Best practices for ensuring the specified system has an controlled amount of 'overengineering' in the tap values to allow for volume controls to sit at approximately the 'two-o'clock' position when delivering the intended SPL
Choosing appropriate headroom in the EPR calculation (p155)
The exam guide suggests 10dB of headroom for speech and 20dB for music. This is simple enough in terms of speech, but 'music' encompasses everything from the highly pre-compressed output of an iPhone with the 'sound normalizer' on; to live orchestral performances in amplified concert halls.
I have shared my designs based on this calculation to several amplifier vendors/manufacturers and been told that 20dB is a frankly ridiculously large value for real-world background music. Still, as I mentioned above, I've sometimes found it to be 'only just enough.'
Does anyone else have additional points of reference for headroom required? Say for an mp3 player output playing rock and pop versus a restaurant lounge stage input where the board is set up for moderate overall compression on the mains output? And is 20dB sufficient for a concert hall, or does full-range sound reinforcement of an acoustic live performance with timpani and brass require 30dB? 40dB?
The only source of data I have been able to find on my own is via software created by participants of the online 'fight against loudness wars.' I can use programs like DR14.meter to assess batches of the prepared mp3 files my employer uses as content and discern the differences between RMS and peak levels across our catalogue. (I have not yet done this.) From what I've read online, these values are typically between 8dB and 14dB for mastered recordings. I would further guess that pop and rock recordings trend toward the lower end of that range.
Will software like DR14.meter give me the information I am looking for? (Have I correctly matched the software to the use-case and in compatible units?)
Am I correct in believing that the findings above suggest a headroom value of 15dB is appropriate for most mastered pop and rock music?
Am I also correct in believing that a compressed 2-mix off a restaurant stage would require something closer to that 20dB value, and that 'proper' sound reinforcement in the pro-audio world requires even greater headroom?
Any advice is appreciated.
Applying the RMS vs peak values yielded by the EPR calculation to 70V tap and amplifier selection (pp155-156)
As shown on the screen capture from Biamp's website in figure 6-12 on page 156, the EPR calculation can be performed twice... once with no headroom to determine RMS and once WITH headroom to determine peak. I assume that tap values must then be calculated using the PEAK figure, as the peak is included in the default ERP calculation we are given, and no language in the chapter suggests omitting it. (Nor does any language suggest how we might make use of the peak wattage value once it has been so calculated, if not for determining the speaker taps.)
However, this implies that after the tap wattages found by PEAK EPR calculations are summed, an amp with a greater-or-equal PEAK wattage should be specified. Unfortunately, 70V amp manufacturers typically only publish RMS wattages on their spec sheets. What's more, it is implied by the manufacturers and vendors that the published RMS rating SHOULD be used when specifying an amp based on summed taps.
I do understand that 70V amps aren't so much meant to provide the wattage they advertise, as they are meant to perform at a 70V peak output down to the IMPEDANCE suggested by the tap wattages in the line... (70.7^2 / wattage and all that...) but why would amp vendors confuse this issue even further by citing 'RMS' where it doesn't belong?
Can anyone explain this discrepancy? What is the best practice here?
Lack of clarity on the reasons we increase the summed tap wattages by 150% before selecting a 70V amplifier as shown in step #7 (p158, top).
On page 158, step #7 tells us to "increase the total tap setting power by a factor of 1.5." No reason is given.
The headroom of the audio signal has already been accounted for in the previous step, so the remaining factors I can imagine are:
1) Accounting for any additional impedance from 70V attenuators, cabling, and perhaps speaker transformer loss itself... and thus ensuring that the amp does not fail due to these 'invisible' impedances from 'outside the calculations.'
2) Allowing for some 'fudge factor' to ensure the volume knobs aren't at 100% when the specified SPL is reached.
Folks on this forum have suggested to me that #1 is at play here. And this makes sense... if it were #2, it wouldn't be listed in the 'Specifying a Power Amplifier for Distributed Audio' section... it would apply to directly-connected systems as well.
Can I get confirmation that this factor of 150% is meant to absorb impedance outside of the EPR calculation brought into play by 70V attenuators, line loss, and/or transformers?
Further to this, I know impedance of a cable is typically very low, and I know that attenuators are lossy. Does this mean I can forego or reduce the 150% calculation if my 70V speaker network has no attenuators? (i.e. volume is controlled using a DSP channel or remote amp channel control?) Or are the speaker transformers themselves enough of an added load to require the 150%?
Best practices for ensuring the specified system has the desired amount of 'overengineering' in the tap values to allow for volume controls to sit at approximately the 'two-o'clock' position when delivering the intended SPL
Finally, and as mentioned briefly in the previous section, we come to the last issue: If I want the system to perform at a nominal level of 72dB-SPL-A at the listener, how can I adjust the calculations to place the 72dB playback level at approximately 2 o'clock on the volume controls when this level is reached? (I mean, obviously I can overengineer the taps and back off the amp trim... or limit the DSP volume range... but neither of these save $$ on my clients' budgets.)
My guess would be that I should perform my EPR calculations with the MAXIMUM possible required SPL instead of the OPTIMAL/STANDARD SPL... but is there any guideline as to how many decibels should be added to put that dial around 2'oclock at a given dB-SPL?
Thanks to any and all who have read this far. I hope to meet you in the comments.
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Christopher,
That's a big topic! We teach all of this in our Design school, it takes several hours to get through it all, but I will try my best to address your questions.
Generally speaking, we want to setup a system that delivers the appropriate level of audio at the listeners ears, with very low distortion and very low noise. Sizing the amplifier properly should give great results.
When developing AV systems, we always start from the human and work back from there. We need to figure out, or assume, a desired dBspl A weighted level at the human's ear. Then apply some headroom so that the amplifier doesn't distort when a higher level signal enters the amplifier. Now we need to find the loss over distance from the loudspeaker to human's ear. The loudspeaker's rated sensitivity and a given distance. Perform the calculation to arrive at your tap setting. Find the sum of all loudspeaker tap settings in the chain. Upsize the total wattage by a 1.5 factor, to make up for the loss in the transformers. Use that final calculation for your amplifier size. One more thing, the signal feeding the amplifier needs to be a voltage that is specified for the amplifier, generally 0dBu or +4dBu (0.755V or 1.23V). Always check your spec sheets.
Let's breakdown your questions individually so if we have follow-up discussions, we don't get a crazy thread going.
Choosing the proper headroom for your program audio.
Simply put, 10dB for voice and 20dB for music. That is VERY simplistic, but are a really good place to start. Running out of headroom is where distortion occurs.
All the things you mentioned are to be taken into account. If you really know your content, you can make more accurate guestimates on the headroom you actually need for your systems. The adage, it depends, is true.
If you run live sound for a symphony, you should expect that there will quiet passages and some very loud passages. What is the total range, in dB, does that represent? Is 20dB out of bounds or do you need more? Are you using compression in your signal chain? Music coming from a streaming channel has been preproduced and most likely has a lot of compression, how much variability is in that stream? Maybe 5dB? If you know the variability, you can use that number, if you don't, it's better to use the 10dB and 20dB numbers to get you started.
If the amplifier provides more power than you need, you are generally better off. Distortion usually happens when you don't have enough power available and the amp struggles to keep up.
If you are getting distortion and you think it you have calculated the system properly, double check the level entering the amplifier is correct. It could be the level is too low and the amplifier can't deliver enough level to the loudspeakers.
Loss over distance and loudspeaker sensitivity.
We must take into account the loss of sound over a given distance. The two points to look at are the reference point of the loudspeaker given on the spec sheet and the listener's ears. Many times the loudspeaker will have a sensitivity rating of x-dB at 1 watt and 1 meter. Meaning if I put 1 watt of energy into a loudspeaker and measured at a distance of 1 meter, I should get a certain level. The formula for the difference is dB = 20log(D1/D2).
If you had a loudspeaker that had a sensitivity rating of 89dBspl @ 1W, 1m, and you had a loss of 4dB coming from the loudspeaker to the listener's ears, the level at the human should be 89dBspl - 4dB = 85dBspl.
If we know the level that we want at the listeners ears, the headroom required, the loss over distance, and the sensitivity of the loudspeaker, we can use the ERP formula to size our taps for each loudspeaker.
In cases where the ceiling is relatively low, the tap setting is almost always the lowest tap setting available. We should do the math anyway.
RMS v Peak levels - which and why?
RMS - is where we sample the signal, square the measurement (converts negative numbers into positive numbers), find the mean or average of the samples, and now find the square root of the number (to undo the squaring but leaves the numbers as positives). Basically averaging the signal, as opposed to finding the peak values.
Average verses peak is really, the normal signal that you want, including headroom. The peaks need to fit into the headroom or you can use peaks to find your headroom. Many digital mixers give you an average meter and peak meter in one. Sometimes known as a peak hold display. This type of display will help ensure that you are not clipping in the mixer or giving you an indication of what you are feeding the amplifier.
What's up with the 1.5 upsizing?
If you calculate an impedance for a loudspeaker chain and then you actually measure the loudspeaker chain, your numbers probably don't match very well. Why? Mostly because of transformers.
When you look at loudspeaker specifications, it is rare to find a spec on the transformer. Every transformer has loss, and we need to account for it. A multiplier of 1.5 is approximately, 10log(1.5/1)=1.8dB difference.
If you didn't keep track of the transformer loss, you could under power your amplifier.
Another factor is the nominal impedance of the loudspeaker is a form of averaging all the impedances at all the frequencies, while your impedance measurement was done at 1kHz.
Over engineering without over engineering
If you have done all the steps, you have properly sized the amplifier.
Here's another way to think about it. If you sized it perfectly and purchase an amplifier that is exactly the calculated size, the volume attenuator on the amp should be set x dB down from max. The x dB should be your headroom.
In the real world, you will purchase an amplifier the next size up from your calculation. The attenuator will be turned down past the headroom level.
Actually, you will set the amplified with an SPL meter at the listening position with program audio.
If you are not getting sufficient level, double check that the source signal to the amplifier is what the amplifier wants to see.
Another thing to consider is the EQ to the loudspeaker chain. Transformers don't pass lower frequencies. If you roll these off, you won't load your amplifier down. This leaves more energy for the frequencies that are required.
I hope this helps.
Thanks for Paul's deep dive posts (and Christopher for bringing topic up) ... just chiming in to also highlight that there was a correction to the EPR formula from CTS-D Exam Guide, Second Edition. You can find that at following https://xchange.avixa.org/manage/posts/196069