Category Archives: Sound Engineering

Feia – cassette restoration case-study

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After a few weeks playing with head alignments, audio interfaces, decks, plugins and sanity, I’ve run off a successful “first draft” attempt to restoring these interesting recordings.

About the cassettes themselves…

The cassettes themselves are a little odd – they appear to be using Type-II (CrO2) shells, but I can’t tell from listening or visual inspection whether the formulation on the tape is actually Type-I (Ferric) or Type-II. Both tapes seemed to sound better with Type-I playback EQ, selected in each case by blocking the tape type holes in the shell with judicious use of Scotch-tape.

Noise levels on the tapes were horrendous. Both cassettes seem to have been recorded about 10dB quieter than most commercial tapes given to me in the same batch, and seem to have experienced significant loss of high-frequencies – something that I noticed getting audibly worse with each playback pass despite cleaning and demagnetising the heads before each run. At best I was getting something like 15dB signal-to-noise before noise reduction. Much of this is broadband noise, but there’s also a significant rolling static crackle running on the right channel, which seems to match the rotational speed either of the pinch-roller on the deck, or perhaps the guide capstans inside the tape shell itself.

Playback

Something I’ve always known about the Akai deck I’ve now inherited and restored to working condition is that it’s always played a little fast. While I’ve not been able to fix this at a hardware level (seems to involve fiddling with the motor control circuits – a major stripdown and rebuild I’m not convinced I have the time or confidence to complete without an accident), I have taken an average of how fast the machine is playing by comparing songs from an assortment of pre-recorded commercial cassettes with digital copies from CD or previews on iTunes. From this I discovered that pulling the playback speed down to 95.75% of the sampled audio gives an acceptable match (within 1 second or so across the side of a cassette) to the commercially-available digital versions. This is really easy to do in my audio software as it doesn’t involve convoluted resampling and slicing to keep the original pitch.

Noise reduction

Challenges

A significant HF-boost was required to get the tape sounding anything like a natural recording, which of course brings the noise levels up. I don’t have access to an external Dolby decoder, and the Akai deck used for doing the transfers sounds very strange with Dolby B engaged even on well-produced pre-recorded material that came to me in excellent condition. The Denon deck I have is technically better than the Akai in many ways, but to beat the Akai in sonic terms needs about an hour spent on alignment (per cassette) and the source material needs to be in excellent condition. So I proceeded to transfer the content from the Akai at a known higher running speed, without Dolby decoding, in the hopes of being able to fix this later in software.

Decoding for playback

There is a lot said online about the mechanics of Dolby B, and many people think it’s a simple fixed 10dB shelving HF EQ boost (emphasis) on recording, that is easily dealt with by a simple shelving HF EQ cut (de-emphasis) on playback – or even simply doing nothing with older tapes that have suffered HF loss. Well, without going into detail that might infringe patents and/or copyright, let me tell you that even from listening to the undecoded audio, it really isn’t that simple. What we’re dealing with here is some form of dynamic processing, dependent on both the incoming frequency content AND the incoming levels. Even with its modest highest-available noise reduction, it’s a beastly-clever system when it works, and remarkably effective in many environments, but as with many complex systems it makes a lot of assumptions, open to a lot of factors influencing the quality of the output.

Working up a solution

Having no access to a known-good hardware decoder that could be calibrated to the tape, I set about using a chain of bundled plugins in my Reaper workstation software to mimic the decoding process. Having been through the process, with hindsight I can see why there are so few software decoders for Dolby B on the market, even without considering the patenting issues surrounding it. It’s a tough gig.

For this process, I picked out the best-sounding pre-recorded tape in our collection and aligned the Denon deck to it, listening for most consistent sound, running speed and dolby decoding.  I got a sound off the cheap ferric formulation that came subjectively very close to the same release on CD or vinyl in terms of listening quality – the tape suffering only slightly with additional HF grain, with some through-printing and background noise evident only when listening at high levels on headphones.

I then aligned the Akai to the same tape before sampling (without Dolby B decoding) and correcting for speed. A rip of the CD, and the samples from the Denon, were used as references as I set about creating the software decoding chain – keeping overall levels the same between reference and working tracks to ensure I was comparing like with like.

A day was spent setting up and tweaking the decoder chain before I came out with a chain that gives equivalent subjective performance to what the Denon deck can do with great source material. I tried the same settings on a variety of cassettes, and was able to repeat the results across all of them…

Content, replication and mastering issues?

…until I came to the content of the Feia tapes I was planning to work on!

Once the cassettes were digitised, and playback speed and overall frequency response corrected, each side of the two tapes was given its own stereo channel, so that individual EQ, channel balancing and stereo-width settings could be assigned to each side of the tape, since I noted some differences in each of these areas that were common to each side of each cassette.

While listening to the digitising run, without playback speed correction, I noted a 50Hz hum in the recordings that was common to all sampled media – I tracked this down to issues with signal grounding between the audio interface, the monitor amplifier, and the cassette deck. No amount of tweaking this signal chain could get rid of it, but with the tapes sounding significantly worse with each playback pass the only way forward was to remove the hum using an FIR/FFT plugin. I therefore set one up on each of the stereo channels and sampled a section of the noise (without the content) into each filter and tweaked the removal settings to be more subtle than default – this removed the hum but left the remaining signal (including bass-notes passing through the hum and its harmonic frequencies) intact.

Each stereo channel was then taken out of the master mix and routed to two more stereo channels – one for the noise-reduction decoder and the other for the side-chain trigger telling the decoder what to do.

Listening to the results at this stage was intriguing. Even after tweaking the decoder threshold levels I noted a general improvement in the signal quality, a reduction in noise levels, but still a strange compression artefact that was evident on high frequencies. This got me wondering whether the labelled Dolby B encoding was actually a mistake, and whether Dolby C had been applied by mistake. Cue another day spent mimicking the Dolby C system by tweaking my homebrew decoding system. Nope – compression still there, but the overall spectral effect of decoding Dolby C was having way too much affect on the mid and high frequencies.

So: onto the next likely candidate: dbx noise reduction. I found out more online about how it works and created an encode/decode chain in software, using a ripped CD track as source material.  Applying the decoding stage to the Feia recordings was dynamically a little better in the top-end, but still not right.

Combining the homebrew Dolby B chain, and following it with a little dynamic expansion on the top 12dB of the recording made a useful difference.  Suddenly transients and sibilants sounded more natural, with more “bite” and less splashiness on the decay, particularly at higher frequencies.

Neither tape is sonic perfection itself even after this restoration, but I’ve learned a lot through it, and how have a much better understanding of why cassettes *can* sound great, but generally don’t, especially recordings made on one deck that are played on another.  I now realise that I’d far rather deal with vinyl and pre-digitised content than extracting it from >20-year-old compact cassettes! At some future point, I’ll likely post up some before/after samples so you can judge the results for yourself.

Appeal for info: Feia cassettes – Circa 1988-1992

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Appeal for info: Feia cassettes - Circa 1988-1992

I have these two cassettes to restore for my grandparents, who have lost their last working cassette-deck to age. The tapes themselves don’t sound to be in great shape, and the claimed Dolby B noise reduction doesn’t seem to play well on any deck I have tried them on. The titles are “Con Amore” and “Canzoni de sempre”.

The tapes shown here were purchased direct from the artist during/after some of her performances in various hotels around Sorrento, during the late 1980′s.

Some questions come to mind:

  • Are these two titles still available for sale, preferably on CD?
  • Anyone else even heard of her?
  • Is she still singing?

UPDATE:  More info from the sleeve notes:

Produced and info by:  P.H. Productions, Marijkestraat 12, 2171 XD, Sassenhiem, The Netherlands / Olanda

I’m getting the impression that this was a small outfit, judging by the lack of a record catalogue number on the cassettes or inlay cards.  This was confirmed by Google Maps, which tells me that the given address is now residential, and looking at the buildings on Streetview suggests this might well have been the case in the 1980′s!

Headphones in live mixing.

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[rant]

I’ve probably posted about this before – it’s certainly one of my pet-hates. 

Image

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,
  • distortion,
  • crackling,
  • 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!

[/rant]

Samson S-Monitor test

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I’ve been building a cabling solution to make a basic IEM (in-ear monitor) system for work, and have had the opportunity to do some critical listening of our chosen headphone amplifier, the discontinued Samson S-Monitor.

It’s a simple device: feed it power from its wall-wart PSU, then feed it a stereo signal to its ‘mix’ input via the unbalanced stereo 6.5mm jack on the back. It can also receive a dynamic microphone input on balanced XLR. Each input has its own level control, which feeds a fixed-level amplifier which can power its two headphone outputs, both on conventional 6.5mm unbalanced jack sockets. I’d prefer for one of these to be a 3.5mm socket so that earbud/canal earphones can be used without an adaptor.

For the experiment I’ve plugged the unit into my EMU 0202 USB, fed from a Linux laptop providing audio from ripped CD’s, upsampled in real-time to 176.4KHz 24-bit.

Compared with the headphone output of my usual hifi amplifier, a NAD 3020B, all headphones have sounded smoother and more controlled when fed by the Samson, with less grain and what feels like a more linear, more dynamic sound at all playback levels.

Used with Sennheiser HD25SP’s, their sound becomes brighter, with wider soundstaging and less of a sense of being ‘closed in’ to my own head.

With Shure SE110 canal earphones, their sound is both brighter in the treble range, with more bass being allowed through and more dynamic range, especially on percussive instruments.

For a laugh, I also tried the earbuds bundled with my iPhone, which took on a much better top-end than when used with the iPhone or the hifi, but became over-warm in low-mids. That said, they did have better control of low bass notes, and seemed to play an octave lower than I’m used to.

All in, i’m surprised at how good this cheap little device sounds! I’ve yet to make it distort before either my eardrums or the headphone drivers give out – which suggests a good amount of headroom and good current delivery.

The only downside seems to be that it’s more sensitive to mobile phone interference than i’d like – If the headphone cable runs against a phone in a pocket, some interference gets through. I couldn’t make it pick up interference when holding an phone to the power cable or the audio input cables.

My advice? Buy one while you can.

Shure and Sennheiser customer service woes

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Been having some issues with some of our mics at work, and some phone calls to the technical teams at Shure and Sennheiser a couple of weeks ago proved helpful in the diagnosis of each mic.

So last week I sent each an email to their published addresses gleaned from their websites to arrange returns codes so their repair teams can put things right at our expense.

Yet, a week on, I’ve still not heard anything. So today I get to make more phone calls at our expense to chase things that have fallen through the net.

If you’re a business, and you bother to publish support email addresses and a process by which to use them, please can you at least respond in a timely manner? Businesses like mine will feel confident and continue to buy your products, paying a little more for the knowledge that you’ll support them. Lose that confidence, and we’ll move to buying cheaper products that end up costing us less both to buy and replace than a single repair process, while still giving us 90% or more of the performance.

Your ball.

MP3 compression ABX test…

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A friend pointed me at this interesting ABX test website that compares MP3 audio compressed at 320Kbps with MP3 audio compressed at 128Kbps.  Comparing A vs B and hearing discernable differences is one thing – but can you blindly hear a track and correctly identify it as A or B?

In the A vs B test, I could hear a difference between the two clips in each test, but there was very little in it.  It would seem that MP3 compression technologies have improved a lot since I last used them with any seriousness.

As for the “blind” test, it turns out that despite today’s tiredness, my use of Sennheiser HD25′s on a standard laptop headphone output, my hearing isn’t so shot after all!

Try the test for yourself here at mp3ornot.com!

Thoughts on mixing sound for Christmas Praise 2010

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Thought I’d dig this older post from the depths of another blog that has yet to go public.  I had the pleasure of setting up and mixing for Christmas Praise last December in All Souls Langham Place. It was a great concert, featuring the All Souls Orchestra with guest stars Michelle Todd and Graham Kendrick.

I freely admit that orchestral music isn’t my strong point, but I enjoy the challenge of understanding and working with something a little different to my normal tastes. This event proved challenging because a real orchestra in this space produces a lot of volume on its own – without amplification of any sort. This means that Michelle and Graham’s vocals needed to be much louder than this PA system would normally be asked to produce, and the orchestral backdrop makes the quality of any sound reinforcement much more critical if we are to blend reinforced vocals with an acoustic orchestra.

Main challenge – Vocal levels

In the event, the quality of vocal reproduction was far less of a challenge than the sheer quantity! It didn’t help that the solo singers were some ways in front of the PA speakers reinforcing them, making gain-before-feedback more of a challenge in this room than usual, especially given the very wide dispersion of the installed Bose 802′s! I was very concerned about tipping the system over into feedback (which would have been a huge distraction) as well as missing cues, so despite a good rehearsal/soundcheck time the first half was spent fighting the system to give me cleaner and louder vocals. I was impressed at having equal numbers of people approach me during the interval saying “great job on the sound”, or “Sounds great where we are but can we have more vocals please?” – that 50%/50% balance is usually difficult enough to achieve in this building with quieter setups!

By the time the second half started, I felt more comfortable with the system as a whole, and therefore set about fixing the vocals that I’d been struggling with for so long. The key to this was making the vocals loud enough to be heard above the orchestra, without having them feed back or go too loud for comfort/comprehension.

A use for compression

The first tool I brought out of the box was compression for the two vocal mics, which I set to the “vocal” preset in the iLive board, with a ratio of 2:1, soft knee, with the threshold set so that the compressor was just beginning to act (around 1dB reduction) on median volumes from each singer. With this, I was able to squash the loudest passages of each singer by about 6dB without audible pumping or feedback issues. This reduction figure was important – it essentially means a halving of the volume. if I could reduce the loudest passages by that much, I could use the “make-up gain” setting to boost the quieter passages by the same amount without having the loudest passages get too loud. Extremely loud passages will get squashed a bit more than 6dB, keeping things comfortable on particularly strong notes or the singer getting right in close to the microphone. This boosting of the quieter passages with control on the louder ones meant that the singers were *always* above the orchestra with no need for me to ride the faders, unless the orchestra also got too loud around them, in which case there’s nothing wrong with reacting by pushing the vocal fader(s) up to restore the balance. The make-up-gain in this case was turned up gradually with care to listen out for feedback creeping into quieter passages.

A use for gating

Given that Graham’s mic was much louder in the singer’s monitor wedge than Michelle’s, his mic was much more susceptible to feedback. The 6dB makeup gain put his mic on the very edge of feedback and was frankly a bit of a liability. I mitigated this by using a gate to take 3dB of gain off his mic whenever he wasn’t singing – 3dB being barely noticeable if I’d got the threshold too high for his quietest passages while I was setting it up, but enough to take the mic out of its perpetual near-feedback zone when he backed away from it.

Other things I did…

Recording

Premier radio came in to record a broadcast mix of the event, for use on Boxing Day I think. I assigned three mono auxes to send an ambient mic feed, a mono mix of the conductor’s mics (more on those in a while) and a mix of everything that was sent directly to the main sound system.

To make sure there was a backup, I decided to try out the M-MADI card in the iLive system and use that to send direct outputs of each incoming mic channel to a laptop. Allen and Heath very kindly arranged for a loan M-MADI card in lieu of any other supplier being able to provide one. With this arranged I was able to hire an RME MADIFACE for use with the one laptop I could find in the building possessing an ExpressCard-34 slot. As I was testing the robustness of the MADI interface as well as that of the laptop and software receiving the audio stream, I enabled all 64 input channels that the iLive system can theoretically cope with to be sent from the iLive, and set up the recording software to stream all 64 channels to disk.

All was well on the night, it seems – having listened to snippets of the 90Gb of audio data we created over the several hours the event lasted, everything seems to be locked together and I’ve yet to hear a dropout. It’s amazing to hear how little noise there is in the recordings compared with the same kind of activity I’ve previously done with analogue gear recording to an Alesis ADAT-HD24.

Live reinforcement

The orchestra certainly packs a punch in this building when fully unleashed – but often the strings and woodwind tend to get lost, drowned out by singers and percussion. To help mitigate this, I put a single condensor mic on a stand in front of the woodwind section, and a cross-pair of condensers in front of the conductor’s stand to get a stereo image of the whole orchestra. This pair was physically closer to the strings than any other instrument, giving me a slightly strings-heavy mix of the orchestra. As well as being used for the recording, i found them useful in the live mix, boosting the strings sections a little so they remained mostly audible even when everything else got loud. I was even able to set them up on their own audio group, so that I could use boost their level the matrix feeds for the speakers covering the sides of the building without having them too loud in the main speakers.

Another useful tool was the channel input delays. Any mic behind the speakers was delayed according to its relative distance from the central point between the main speakers using the 1ms-per-foot rule, the approximate speed of sound travelling in the air. This meant that drum and piano mics in particular effectively disappeared from the mix as identifiable sources, so i could mix in a little more of these instruments for clarity and impact without them sounding like these details were being provided by the sound system. These delays were also recorded to the hard disc, so the live recording immediately sounds more natural when played on headphones or a decent hifi system.

Summing up

This was not the best-sounding mix I’ve ever created for an event, but the tools built into our iLive console certainly helped bring things under control with a whole lot less stress and anxiety than I’ve experienced with similar events in this church using the previous analogue console. The ability to stream every input to its own channel in a software recording system certainly made reviewing the work after the concert a whole lot more meaningful, so I’ve now got a list of things I’d do differently next time I run a similar event. More on that note in a future post, I think!

Audiophile or Audiophool?

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I’ve been getting way beyond my usual audiophile tendencies in the last month or so, and was brought back down to Earth by a forum post pointing me to this cartoon on xkcd:

Sums the situation up quite nicely, especially as I struggle so much with understanding what I hear over a mobile telephone connection!

Simplicity is sometimes best…

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Just back from a wedding of two dear friends – itself a great day.  I was humbled to have been asked to contribute to their service by doing live sound duty, on an unknown PA system in an unknown church.

I arrived at what I think was the most simple and effective PA system I’ve yet encountered in a small church:

  • 4 x EV SX300′s
  • 1 x Behringer power amplifier (not sure which one)
  • 1 x Allen & Heath GL2400 analogue mixing console
  • Mostly fed with Shure dynamic microphones (even for lectern-based speech!) or passive DI’s
  • Mackie SRM450′s as stage monitors

It has to be said that I didn’t expect miracles – this is to some engineers the most basic system they’ll encounter.  No need for delay or dynamics protection for the speakers, no graphic EQ even for either main or monitor feeds.  So what went into the mixing board was going straight to the speakers unaffected.

What struck me was how “direct” the mix felt.  Though I’ve used systems that felt more revealing in terms of minor EQ tweaks and such, the system punched well above its weight in terms of clarity and dynamics.  The SX300′s aren’t the best speakers in the world, I know, but the sound they gave was a convincing reproduction of what I knew to be coming into the mics.

The other significant thing about today’s task was that I’ve spent the last three or so months of my working life configuring and mixing with an entirely digital console.  So of all the analogue boards I could have encountered it was refreshing to find myself so at home with the GL2400.  This is a good, no-nonsense board that I felt immediately at home with.  It is equivalent in concept to the Soundcraft Live 8 series consoles, I guess.  Combined with the simple amp/speaker setup, I felt completely at home with the board, and it was a real joy to mix with.

So – Well done St John’s Stoke (Guildford) – you’ve managed to pull off the impossible.  You have a system that is technically adequate, that sounds good and that has minimal visual impact on your building.  Further, the in-house tech’s I spoke with were friendly, willing and knowledgable – which are always encouraging qualities to encounter.  I wish you well in your ministry, and despite working in a venue that has much more complex needs to deal with, I could learn a lot from your approach!  So thank you.

Acoustics Experiment 2 – Main sound system overview

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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:

Sound mix position (802/302 only)

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.

Centre aisle, front row level (802/302 only)

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:

Central Aisle, Lectern mic position (802/302 only)

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.

Signing off…

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.