I’ve probably posted about this before – it’s certainly one of my pet-hates.
I often find myself wanting (or having) to remind noise-boys happily mixing in their own head that they’re ripping the heads off the front 20 rows because they’re not dealing with an EQ, balance or feedback issue that they’re not hearing!
In summary, headphones are a great way to check individual inputs or outputs for:
problems with mic positioning,
presence of signal,
balance of a mix YOU CAN’T HEAR IN THE ROOM, such as those for foldback, recording, or other rooms.
When NOT to use headphones… In live sound reinforcement, headphones are simply terrible for listening to the mix that you’re producing in the room, whether for judging mix levels or tonal quality. You’re hearing small speakers, close to your eardrums, mixing for a space that exists only in your own head.
So don’t use headphones to check or perfect your main mix – it’s truly an accident waiting to happen.
If you’re changing something and you can’t hear a difference in the room then going to headphones isn’t going to help anyone hear that change except you. Worse still, while you’re dealing with a problem that only you know/care about, you’ll likely miss something crucial that everyone will notice!
Those who have read my little review of the Tannoy Mercury M20 Golds might be aware I’ve inherited some other items that they were originally purchased with sometime in 1985, we think. In this little article I’ll be explaining a little about our (immediately) beloved turntable – the Dual 505-2. By modern standards, if purchased new I would guess this record deck would compare with the likes of the Project Debut Essential package or similar. Our example appears to still be fitted with its original Dual ULM165E cartridge and DN165E stylus. We have no idea what playing time the stylus has seen, nor whether it is indeed the original stylus or an after-market replacement.
Fault-finding and repair
Upon arrival the platter would not spin, and I had been warned of the need for replacement drive and pitch-adjustment belts. After some Google abuse in an attempt to find a service or owners’ manual for this unit, I found my way to the Vinyl Engine which had an English owner’s manual available.
Dual 505-2 disassembly/reassembly
After downloading and reading the owners’ manual, I then attempted to take the inner plinth out of the wooden frame. This isn’t as easy as it looks; so having armed yourself with a Leatherman, Dutch courage and a glance at the owner’s manual, it goes something like:
Lock the tonearm in place.
Remove the stylus and put it somewhere safe to save it getting damaged.
Remove the rubber turntable mat, and the platter.
Remove the plastic lid.
Ensure the transit screws are in the “playing” position, to give the suspension mounts full movement.
Turn the whole deck on its back (so it rests with the hinges against your work surface).
Slide the suspension spring bases out of their homes in the plastic base plate. The whole plinth assembly can then separate from the base. Note that the captive mains, signal and ground cables prevent the plinth separating completely from the base, without further work to release the cable entry glands.
Reassembly is essentially a reversal of steps 1>8.
Checking the tonearm, motor and microswitch interaction
Another thing I learned during my Google session to find the manual was that one of the most common faults with these decks is that the microswitch seizes up if the deck has not been used for a while.This is the switch that starts the motor spinning when the tonearm is moved into the playing position. The cure recommended on most online forum posts I found on the issue was to simply use a screwdriver to operate the switch enough times until it starts working again. If I recall correctly, the switch is on the underside of the plinth next to the motor, and has either a yellow or blue plastic cap that connects to the tonearm assembly, via a system of levers that I could not easily work out. Within about 10 pushes the switch mechanism freed itself and the motor started spinning. With hindsight it was risky leaving the mains plugged in while I took everything apart, but it paid off and as it turns out there were no exposed terminals that a stray finger or screwdriver could have found. Phew.
As mentioned earlier, I had been warned of the need for a new drivebelt. It turned out that the belt itself was fine, but it needed a little gentle persuasion to realign it so it ran inside the speed selection mechanism. I tested the speed selector a couple of times to check that the belt stayed in the correct position, which it did. While I had the deck apart I also discovered that the toothed pitch adjustment belt had somehow snapped, so without any spare parts to hand I simply removed it and hoped for the best.
Function test, and adjust pitch
Having reassembled the deck I plunked down Suzanne Vega’s debut LP, and found both platter and cue mechanisms to work as designed. The result was quite stunning – the aged stylus and cartridge combination was working well enough with the photo stage of my NAD 302 amplifier to extract a remarkably pleasant sound from the disc, albeit at a slightly higher pitch than normal.
So, off came the platter and out came the Leatherman to attempt a quick-and-dirty adjustment of the pitch control pulley. Having worked out that the belt links the surface-mounted pitch control knob to the control pulley on the motor assembly, I could reason as to which way to turn the pulley to make the required adjustment. Using the narrow flat-blade screwdriver on the Leatherman, I turned the pulley through 90 degrees anti-clockwise by locking the blade in the teeth of the pulley and pushing gently in the right direction.
Trying the test LP again showed I’d adjusted too much – the song was now playing slightly slower than normal, so I went back and halved the difference. The LP now played what I considered to be ‘normal speed’ (turns out I do have an intuitive sense of ‘perfect pitch’, though it helped to have a CD of the same album to compare, which did indeed synchronise perfectly both in terms of track length and perceived pitch/tempo.
First impressions of sound quality
This is a subject for another post of its own, I’m sure. What immediately strikes is that the sound quality on offer is surprisingly good, but there are no good words or phrases I can think of to describe how it differs to the same material played from CD or other known digital sources in my system. When the disc is clean and in good playing condition, and has itself been mastered and manufactured well, the soundstage is noticeably wider and deeper than my digital sources, and the overall presentation is simply more musical. It’s not that the vinyl source is ‘warmer’ or more detailed as has become the stereotypical wording used by audiophiles writing in print or online when selling the plus-points of vinyl – just that the overall result is simply more pleasing to me. My wife confirmed this, noting that the vinyl feels more ‘real’, more as if the musicians are being presented in a space around and in front of us, compared with the digital sources forcing the soundstage to be artificially contained within our room.
Stay tuned for more on what work we carry put on this deck, and for some more in-depth reviews of what it enables us to enjoy!
Put picture back into the post as originally designed, correct spelling/typo’s, add the following list:
Things still to do:
Install a replacement stylus in case I broke anything in transport or handling. Turns out the current one (and its cartridge) are replacements with only 50hours playing time on them, but I’ve already ordered a Dual DN-165E replacement stylus from Stylus Plus, so at least I’ll have a spare.
Replace the pitch control belt – not because it’s needed in normal operation, but I feel it’s only fitting to bring this one back “to spec”.
Check alignment of the cartridge. When I first started using the deck, I noticed a significant amount of sibilant distortion when nearing the end of a side. This isn’t uncommon, but a quick alignment according to the “Stevenson Method” has made things noticeably better at the expense of slightly increased sibilant distortion at the beginning of each side.
Clean all our LP’s and replace inner sleeves.
Try the phono stage of our inherited NAD 3020B in place of the current NAD 302.
Based on the system EQ settings (and process) that inspired this blog entry a couple of weeks ago, I took some time to do some system measurements to see what kind of pink-noise response the system gives now it’s been tuned “by ear” in our building. This measurement session also set the groundwork for another blog entry concerning the spillover from stage monitors into the rest of our building.
I freely admit that these measurements were taken from mere curiosity and without any specific question in mind, nor were they taken with any specific point to investigate. For this reason I find the measurements and their interpretation so interesting.
Theory of RTA measurement in sound systems
For true reproduction of a pink noise source, one would expect the 1/3-octave bars in the RTA display to all be at the same level – pink noise has the unique property of having equal power per musical octave. I would suggest that this means there is as much power in the range from 20-40Hz as is found from 2000-4000Hz – hence explaining the “flat” response that should be seen on a 1/3-octave RTA such as my iPhone app used for these measurements. Further reading on the theory of pink noise can be found in this wikipedia article on the subject.
A “perfect” sound system
I think the most commonly accepted definition of a “perfect” sound reinforcement or playback system is that it plays back exactly what is fed into it. This definition seems to hold true in most fields, at both consumer and professional levels. The thinking here is that if you play pink noise into the sound system, you should observe an exact replica of that original signal coming out of the speakers. Any tonal (frequency-based) deviation from that original signal will show up on a 1/3-octave RTA display as a peak or a trough relative to the the neighbouring bars.
This is the basis on which EQ’ing/analysing with a pink-noise source is built – we know the source, and in theory we can therefore shape the frequency response of the system to get the height of the bars as even as possible. Having made suitable adjustments to the sound system EQ and speaker choice/positioning, we ought to be able to draw a horizontal line from the top of the lowest frequency bar to the top of the highest, with no one bar falling above or below that line.
Of course the reality of many sound systems, even the ones we would judge to be “excellent” often fall quite short of this aim, due to a combination of physical system/room parameters and interactions, as well as the intentional “voicing” of a system by a human engineer/operator to flatter its usual source material. Humans often like to interpret things in this quite non-scientific way and when this is done well it is considered an artistic addition to the system. When such deviations are present due to the nature of the room or the system itself, or if they are due to an inappropriate “voicing” by a human engineer, these artefacts tend to be considered bad things.
On with the measurements!
Sound mix position
With all the above in mind, I thought it interesting to measure the output of our system when fed a pink noise source, after the system was EQ’d by ear to a series of test-tones, where I intentionally made a perceptive judgement of the apparent loudness of each tone relative to the others. Since these judgements were taken from the sound mix position, I thought this a good location from which to take my first reading, shown here:
Now, this graph is interesting. Well okay, *I* find it interesting, even if nobody else does! Overall, it suggests a reasonably even response from 20Hz up to around 2KHz, before the response of the system falls off significantly at around 4KHz, before climbing steadily up to 16KHz. Above this point, either the sound system or the measurement mic seem to show a reduction in output. I’m not terribly concerned about anything above 16KHz because many people cannot hear much above this anyway, and those who can will tend to not be too bothered by a slight reduction in output here.
But what about that dip centered around 4KHz?
A good question. Human hearing is an odd thing, especially in buildings when listening to abstract tones that bear little relation to real music or human voices. Frequencies have different perceived loudness compared to others, even if a perfect sound pressure measurement system shows them to be played at the same frequency. The human ear tends to be most sensitive to mid-range frequencies, from around 1KHz up to around 5-6KHz. This contains much of the intelligibility components of the human voice, so in some ways it makes sense that our ears are tuned to be most sensitive at such crucial frequencies.
BUT: here’s another thing – our hearing sensitivity changes depending on the volume of the sound we’re hearing and/or responding to. Typical speech levels tend to be in the 50-70dB range in normal conversation in a reasonably quiet room, so I chose 65dB(A) as reference sound pressure level for the test-tones as well as for the pink noise analysis, given that it’s towards the top end of the volume range most people find comfortable when listening to reinforced sound. Any louder and people feel like they’re being shouted at. Any quieter and the effects of tonal inconsistencies tend to be perceived as being less of an issue to the average person.
So what has happened here during my EQ session is that I’ve intentionally pulled a chunk out of the system’s sensitivity to these critical mid-frequencies based on the fact that they seemed so much louder to me than the others, despite their measured sound level being within 1 or 2 dB(A) of those I didn’t find so bothersome. On its own, this is usually a bad move for setting up live sound systems for either speech or music reinforcement, so I checked out the new EQ profile by playing some well-known and much-loved music through the system, and found that the EQ curve still worked. Comparing that to the same music played through a curve without the drop in the 2-8KHz frequency range made the music sound rather harsh and shouty, even brittle somehow. So despite a significant issue showing up on paper, I left the new curve in place on the basis that it sounds pretty reasonable with music and speech playback without any further work being required by the mix engineer to make that material sound good.
Phew – so far so good.
Front row position (seats closest to the stage/chancel)
With my curiosity still ablaze, I took another measurement of the system from the front row of our main downstairs space, using the same pink noise signal at the same input level, and leaving the all other controls alone. This will show up any difference between what is heard at the mix position and what is heard at the front row location.
First off, I note that the measured sound levels at this position are really not appreciably different to those experienced at the mix position – 0.3dB as shown between the 10-second measurement periods at each position. Let’s consider for a moment that most people (even many sound engineers) would have a real struggle to reliably notice a 1dB difference between two signal levels. Essentially we’re saying that someone sat at the front of the venue shouldn’t be hearing a signal level difference compared to the same person sitting at the back of the venue, at the mix position. That’s quite astonishing, and shows something of good sound system design that the coverage is so even.
The second thing I notice is about the content of that sound level. The output at frequencies below 2.5KHz shows to be generally slightly lower than measured at the mix position, but frequencies above 2.5KHz are played slightly higher than at the mix position. Thus, despite the overall levels being so evenly matched, from the measured differences one could expect the system will sound tonally quite different in the front row seats compared to the back.
From experience I can tell you: it really does! I now know that if I’m mixing for “hifi sparkle” (with lots of high-frequency detail) as heard at the mix position, the seats in front of me will receive a mix that is unpleasantly biased towards higher-frequencies, sounding more like the brittle and shouty system that the aforementioned measures-flat-on-a-meter EQ method gave us. Ugh.
The sound system as heard by the most important microphone feeding it
An interesting one this. I’ve often observed that I can hear a whole lot of the main sound system output at the main speaking position we use, and I’ve been telling less experienced readers/service leaders for years not to let this fool them into thinking they can speak more quietly and let the sound system do all the work.
Well now I finally have something approximating proof of what I thought I’ve been hearing for all these years:
Again, there are two things I would note from this graph. The first is that we’re only approximately 1dB(A) down in overall output volume from the sound system at the microphone position than we would be at the back of the church. It’s almost no better than having the microphone directly in front of one or more of the speakers its output is fed into. Gain-before-feedback is a significant area of struggle in our building and with this information I can see why.
Secondly, let’s think about the content of that sound. Again, the output is quite even up to around 3KHz, with less of a relative dip at 4KHz than in either of the measured positions where sound output from the system is desired. In this position we actually don’t want the sound system to be contributing any significant output – and in our current setup we have the very opposite of that desire! A “flat” system response to pink noise would result in a microphone that is already very sensitive to frequencies between 2KHz and 8KHz receiving a lot of its own sound from the system at those very frequencies – leading to vast quantities of feedback if the person speaking into it is delivering less than around 65dB. Sadly for us, many of our less experienced people using the lectern tend to deliver less than 60dB to the mic, so getting them sounding both loud enough and pleasant enough to listen to is a significant challenge.
I hope that rather than bemoaning an ongoing struggle I’ve actually contributed some useful thought and input to the subject that others can either learn from or correct me on. This has been a wonderful learning exercise for me, and I hope the findings can eventually lead to some significant improvements to our systems, our methods and eventually the sound that is heard in our venue.
Last Friday evening I found myself again having to EQ our main sound system at work, due to a combination of what I believe to be environmental factors and physical changes to the speaker setup, namely the replacement of some faulty speaker drivers in a bass cabinet that needed taking into account – itself a blog subject for another time I’m sure!
My usual experience with setting system EQ has usually been centred around one of two things:
Making the system sound as good as possible with CD-sourced playback material, in the hope that this will provide a known starting point for the sound of any mix we create on said system during a live event, or…
…putting a key microphone into its usual position (such as a lectern for a church) and having someone speak into it, making their voice sound as natural as possible (without resorting to desk EQ beyond a simple high-pass-filter). Once this is done I’d then slowly turn up the gain for that microphone (keeping the fader level constant) and using some form of EQ to pull out any frequency bands that feed back.
Both methods have been “good enough” for rock-n-rolling into a venue and making something more than reasonable come out of the speakers, but neither method is terribly scientific, nor does it lead to consistent results.
More recently I’ve been playing with using pink noise and an RTA to show me what the system’s doing, then EQ’ing the system so that pink noise played out of it and measured with a flat-response microphone is shown on the RTA as being as close to the original pink noise as I can get. This has lead to more consistent results than either of my previous methods, and has cut down the time spent on the task by something like 50-70%, but still the resulting system sound is somewhat variable to say the least.
So at a pinch on Friday evening, I happened upon what seemed to be a better method, and one I’ve not tried since my earliest days of sound mixing/system engineering:
Make a CD with test-tones at a fixed level (usually -20dB), centered at the typical frequencies found on the faders of a 31-band (1/3 octave) graphic equaliser.
Starting with 40Hz (the lowest audible frequency in most mixes/systems I deal with), get the tone playing through the system at about 65dB on a typical SPL meter. A or C weighting doesn’t matter at this point – what does matter is that I get it set in my mind how “loud” that tone sounds/feels.
Then go to the next tone up.
Is this playing at the same perceived level as the 40Hz tone that preceded it?
If yes, move on to the next tone.
If no, then set the system EQ (I had both parametric and graphics to hand) to compensate. Keep comparing and adjusting until the level sounds comparable.
Repeat steps 3-4 until all frequencies that you can hear, either due to the system itself or your hearing (!), are pretty much perceptively even.
Note 1: If you have a parametric EQ, and you find that frequencies progressively become more or less prominent than those preceding them, you can set an EQ curve centered at the point where the smallest difference occurs between adjoining tones, boosting or cutting accordingly. The width of the filter is roughly defined by the number of tones you find to be different. It’s hard to explain in text, but becomes more obvious the more you play with the EQ parameters. Using a parametric EQ here gives more precision and control over what you do to the signal, without the distortions of graphic EQ, which essentially is a chain of 31 or more audio filters run in series.
Note 2: This is best done as an iterative process, so it’s worth playing through the test tones up and down the scale and adjusting until you feel you can’t make any more positive adjustments.
Note 3: On our system, I was able to accurately do this up to around 16KHz, as with the combination of my hearing and our system I wasn’t able to discern anything beyond around 17KHz. Not bad for a tired near-30-year-old engineer, working late at night on a combination of Bose 802/402/302’s!
Having applied this method to our system I played a couple of favourite songs through it from my laptop, which has a pretty good quality sound output (equivalent to most “hifi” grade CD players when fed with CD-quality content), and the system sounded immediately more musical, more involving and less “PA-like”.
Out of interest I measured the pink-noise response of our system What I found with this method was that my system curve had a notable reduction in the 2-6KHz range than would be obtained by using the pink-noise method above, which might be seen by many engineers as a significant disadvantage.
I then had a couple of our other engineers use the system in live services with this new system EQ curve, and their feedback was that the system sounded so much better than they’ve been used to. They were making much more subtle (And arguably more accurate) changes to desk channel EQ for both speech and music, and the usual issues we have with feedback or tinny-sounding speech microphones were much reduced.
On reflection, I wonder whether part of the success story here is that my chosen reference level of 65dB (SPL, A-weighted) is pretty close to someone talking passionately to another in a quiet lounge – and given that the sensitivity of human hearing at specific frequencies changes depending on the overall sound level, this coincides quite nicely with our main material, speech reinforcement with some louder music that doesn’t often get much louder than 90dB.
I’m sure I’ve done many things wrong by working this way, but it was quick, easy and seems to have worked out well for us – our engineers are happier working with the system set up this way than they have been for a long, long time. I’m sure I’ve missed a few crucial things out in my explanation here, so I might re-visit the topic in the future. But meanwhile I hope this stands as a demonstration of another way of using EQ to get more out of your sound system.
As always, your mileage may vary – and your needs might be very different to ours!
I’ve just been unexpectedly covering sound duty for our two morning services, and took some time to really play with the EQ facilities offered by our new desk, particularly during the sermon. Above is a picture of the EQ section for the radio-mic I was using, and it looks pretty extreme, huh? Lots of huge cuts if the gain indicators (the red LED’s at the bottom) are anything to go by.
Now, if someone showed me their analogue desk and I saw a channel EQ with quite so much taken out as shown, I’d have taken them back on for more teaching about gain structure and microphone choice/placement among other things. On most analogue desks there’d be nothing left of the original signal. You’d end up with an EQ curve looking a bit like the yellow line shown here:
Extreme EQ, “standard” analogue desk style. Low shelf frequency is around an octave higher here than most analogue desk EQ, and high shelf around two octaves lower. The yellow line goes off the lower scale, it’s so extreme. You’d need to add at least 12dB of gain to the fader or pre-amp to get the original signal energy level back, assuming you don’t distort the pre-amp or other areas of the desk!
On our iLive system, things are a little different, as shown in the image below. We now have the ability to notch out problem frequencies with much more precision, mostly because we now have the ability to create very narrow (in terms of the frequency range affected) EQ filters. For live sound, this means we can make deep, narrow cuts to problem (resonant) frequencies and leave the rest of the signal alone. This means that the problem frequency bands can still be attenuated, but without losing anywhere near as much of the overall signal energy.
This allows us to run with less gain at the pre-amp stage, which makes for less background noise and less chance of distorting any audio stage in the desk, whether digital or analogue. Because I’m not having to boost the pre-amp gain, I’m not changing the gain structure in any way, which is a Good Thing™ for too many reasons to detail here. Because I’m not boosting either overall levels or particular frequency bands, I’m not introducing new potential feedback points to my mix – again a Good Thing™.
Of course, this benefit isn’t unique to the iLive system, but I used it to illustrate the problem as a) I have one and b) I happen to like how it sounds!