[noname]

@DaftFader 460579 wrote:

I never clip my faders … I ment when running stuff thru vst’s … would it matter if they were all red lining at 32bit if the final faders were not clipping, or have I totaly misunderstood that part of 32 bit floating point?

Yes it will matter – if you are redlining any part of the signal chain it will distort whether your mixer is digital or analogue. As I say, 32 bit floating point is just a way of representing in numbers the audio you are working with. If you redline the mixer channel with a VST instrument you are driving it up to a point where there isn’t enough space at the upper end to represent it properly – the result will be clipped audio and distortion (which can obviously be introduced purposefully for a particular audio effect – if you then ensure the main out channel isn’t overdriven it will sound fine assuming you want that overdriven sound for that particular VST – I would suggest you would be better off using an overdrive effect instead of doing this, but if it gets the sound you want doing it this way then it’s all good :wink:)…

@DaftFader 460579 wrote:

See you need to cheet even more and steal pics from google 😛

Normally I would’ve, but I couldn’t find one that showed what I was trying to explain properly :crazy_dru

[noname]

Quite surprised (and chuffed) here – I thought my hearing was fecked in one ear and deteriorating in the other (a result of bad in-club rigs 20 years ago – I remember noticing I couldn’t hear as much when mixing a band one time, and put it down to distorted nasty speakers that I had been working with the year before – I started wearing ear defenders whenever playing in a place that I didn’t have control of the rig in)…

The surprise is from the fact I could not only hear the 17.4KHz tone clearly (and I agree it is very irritating when heard for a short time – I’d likely be fit to bite something if subjected to it for any length of time), but on the hearing test tone page I can hear the tone coming in at 22KHz in my right ear (the not fecked one), and just over 18KHz in my left ear using a pair of sennheiser headphones and setting the sample rate on my sound card to 48KHz. Interestingly when it was set at 44.1KHz I couldn’t hear the tone in my right ear until it hit 20KHz – I assume this is probably down to the DAC on my sound card not rendering the 22KHz tone properly when it’s the extreme top end of the range as it is with the sample rate at 44.1KHz…

Seems the damage did get done to my left ear (which I tended to use to check whether the speakers were working when using in-club systems, so it bore the brunt of the nasty distorted sound), but my hearing is deteriorating nowhere near as badly as I had thought it was raaaraaaraaa…

[noname]

@DaftFader 458349 wrote:

does floating point (32 bit float) mean that you can clip in the digital domain and as long as the mixer faders aren’t clipping it’ll not effect the sound? (I read that somewhere but am not sure how true it is).

32 bit float just means the resolution is 32bit and uses floating point representation (in accordance with IEEE 754 ).

As it’s just a way for computers to represent information, and perform calculations, and doesn’t relate to a physical property of the audio I would suggest that it would have no effect on clipping (which is a physical property of the audio). If it’s clipping in the digital representation then it will be clipping when that representation is converted back into the audio represented…

@dubstep_joe 458487 wrote:

How long did it take you to write that?

A wee while (memory trawling gets more difficult the older you get – especially if your youth was properly mis-spent 😉 )

@DaftFader 458349 wrote:

You should of cheated used loads of pictures like I did.

I tried, but the single picture there took me about as long to draw as it took to write the text :laugh_at:…:crazy_fre

[noname]

@General Lighting 455326 wrote:

OK here’s whats inside it.. its is just two proper audio frequency transformers on a decent bit of PCB and shielded (was pleasantly surprised, expecting far worse. Ok the build quality is your average Chinese standard (though neater than what I would have come up with) – and it worked perfectly after I put it all together again.

Apologies for the colour balance being all over the place on these photos, was rushing a bit trying to get something useable online..

and this is how it all fits together in the box (note the metal screening box to keep out radiated interference).

I have no idea what the transformers are / their specifications but I’m old enough to remember when all hi-fi equipment had many more audio frequency transformers (this practice only fell out of favour by the late 1970s). They do not appear to be filtering excessively as I am currently feeding a FM audio signal into a soundcard via one of these and the 19KHZ stereo pilot tone is getting through…

Daftfader is correct inasmuch that there will be an insertion loss and perhaps change in frequency response, but It all sounds OK to me, and I don’t have the brains to do the proper test (even though that italian software should do it – it hurt my brain enough just getting it to work!)

What I expect it does do is present high impedance to frequencies well over 30 kHZ (including all the nasties from the computer soundcard) which isn’t a bad thing as thats the noise you want to get rid of!

Frankly its a bargain for what you get. In the old days folk would have built one inside a tobacco tin, but its hard to get both baccy in tins and decent AF transformers and my old mate what told me about this was in addenbrookes for some time with throat cancer :yakk: (he recovered though)

Looks awfully like the boxes I’m putting in the car when I eventually install the bass bins and amp – I’m using 2 to go from the head unit to the power amp in the boot (setting them up back to back at either end of the cable essentially gives us a balanced line between the 2 devices – doesn’t need an “earth” point like a normal sound system as the car doesn’t have a standard earth like a house or genny etc, it uses a negative grounding point system…)

No sure how well they’re gonna work yet (I’ll post here when we eventually get round to it), but I’m told they work really well like that (transformer balancing works as well as if not better than servo etc IMO – most of the reason it doesn’t get used much is the transformers are usually pretty expensive especially when compared to an SMD servo chip)…

[noname]

@DaftFader 455297 wrote:

Ah ok, I read that it has some kind of filter (LP,HP and/or BP/BR within it iirc) … and assumed it was placed in the signal path. I guess as this could apply to a general electronic signal, that they were purely refering to blocking electronic frequencies as opposed to specific sound waves (I know they are the same thing esentualy … hence the confusion).

An inductor is a filter – inductor placed inline with a speaker = basic low pass filter (or 1st order crossover).

Inductor of the right value placed on a possibly dodgy ground line stops it from allowing any hum etc through…

[noname]

Aye – pretty much what cheeseweasel and Daftfader said. When I talked about having to use anti aliasing when you downsample – ie from 48K to 44.1K or such it is for the reason explained above (anti aliasing attempts to remove any aliasing errors by filtering everything above the Nyquist limit) Another process called interpolation is used to make the waveform more closely follow the higher rate waveform by altering the placing of some sample points in this manner:

[ATTACH=CONFIG]81023[/ATTACH]

Not 100% sure, but I believe this is still covered in the Nyquist theorem, although it’s called Shannon-Whittaker interpolation after some of the other folk who are responsible for the theory (that only Nyquist seems to generally get his name on for some reason)… Much of the original mathematical proofs are written by Claude Shannon, and Whittaker came up with the same idea independently of them.

Anyways to explain the diagram:

The absolute path for a halfed sample rate (the dotted red line) is much less dynamic than the original at the higher rate – therefore we move some points (the green line) to more faithfully represent the source (obviously it isn’t perfect, but the green line is markedly more dynamic than the dotted red where the signal fails to get out of negative Voltages for 6 samples in the middle – the original does so 3 times during the same period, and the green manages it twice).

Thats a very basic explanation of the theory. The actual process of anti aliasing and downsampling usually involves running a low pass filter to filter out any frequency changes that are above the Nyquist limit (in the same way as converting from analogue to digital – although a band pass is sometimes used in AD conversion because the analogue source is continuous and band pass can help avoid some quantisation errors. Downsampling can’t use a band pass filter, and must use low pass (AFAIK – I could be wrong here though – there are bits of Nyquist that are just too much maths for my brain to cope with…)

Then I believe some form of inverse Fourier is used to try and rebuild the original source as far as possible, and this is then quantised back to the digital waveform required (this could be complete nonsense though – as I say, maths makes my head melt – especially when greek letters and such are involved :crazy:)

As for bit depth, it dictates the resolution of the representation – 8 bit gives 512 levels of resolution, 16 bit 65,536, and 24 bit 16,700,000 or so. This dictates the dynamic range (the difference between the smallest and largest sound representable), and as a consequence of that it also dictates the signal-noise ratio. When audio is quantised (the process of squaring the waveform – necessary because it’s being represented by a series of discrete points instead of a continuous wave as it is as an analogue signal), it introduces repeated quantisation errors (which manifest as harmonic distortion at the 1st and subsequent harmonic frequencies – not generally very nice sounding – think someone planking a flat palm on a piano keyboard and hitting 6 or 7 keys together). So the process of “dithering” tends to be used (where the repeated errors in the quantisation function are randomised so removing the build up around the harmonics).

dubstep_joe wrote:

I’m curious guys. Does anyone know how a sound waveform is encoded? What does the ’rounding’ error relate to? Is the ‘amplitude’ being stored as a 16/24/32 bit integer? Would a bigger bit-size just mean one can encode ‘louder/bigger’ amplitudes?

Rounding error is the afore mentioned quantisation problem.

In PCM (which is what is generally used to encode a wave), the audio is encoded with a set rate (the sampling frequency), and a measurement of the amplitude is taken. This is then quantised into a digital representation (a process otherwise known as modulation).To reconstruct the audio it must be demodulated (using Nyquist/Shannon’s method of reconstructing the original wave, and passing it through some filters to remove the unwanted additions introduced as an unavoidable side effect). This produces a voltage (or current sometimes – depends on the DAC) which is then used as the original analogue electrical source would have been.

The amplitude is stored as a (probably integer) value usually (although from the POV of information, floating point values can represent the same quantity of info and therefore the same resolution). Don’t get confused between the actual physical property of the wave encoded, and our representation of it using numbers – louder is a physical property, whereas the amplitude stored in the numbers is a representation of it (the bit depth allows us to represent it more accurately, and also allows us to change the original, including the amplitude if we have sufficient resolution and the original signal has enough headroom – the “normalise” function in audio progs does this as a way to maximise the dynamic range by increasing the amplitude of the loudest part to 0dB (or the highest amplitude encodable), and changing the other amplitudes by the same amount through the whole wave).

It wouldn’t matter if we had 1 as the highest amplitude – 24 bit would still break the range 0-1 into 16,700,000 discrete values, and similarly if we had 512 as the highest, 8 bit would still only encode 512 discrete values (and in the previous example for 0-1, 8 bit would have 512 values between 0 and 1…) I would imagine most systems use integers (possibly with a single bit for positive and negative designation so the standard representation of a wave oscillating around the 0 point can be imagined easily), because computers are historically better with integers than they are with floating point maths (way back when, before the PC had developed the math co-processor to handle floating point calculations, PC’s were lamentably slow at anything involving floating points…)

Hope that all makes some sort of sense – gonna stop now cos it got a lot longer than I planned it to be… :sign0020:raaa:lol_crash

[noname]

@cheeseweasel 458272 wrote:

I’ve often wondered what an MP3 would sound like to a dog or cat (in fact I remember asking this question in a lecture once and people thought I was taking the piss), as the way MP3 works is by exploiting redundancy in the human auditory system by using the masking effect. This requires a virtual model of the human ear to be incorporated into the encoder, which tells it which frequencies will be masked by the programme material (these frequencies are then re-sampled at a much lower bit-depth but the resulting quantising noise is masked). To a non-human listener it must sound like total junk.

Aye – I would imagine they sound pretty horrible (if you play them on a half decent size and quality rig that so called redundancy in the human auditory system becomes immediately obvious to most people)… They don’t encode anything over 16KHz (the last band of the 32 bank PQF only goes up to 16KHz as the “average” persons hearing doesn’t usually hear much over that). Cats, dogs and the rats mentioned probably hear well into the middle 20-30KHz range and maybe even further, so to them it’s missing a lot of it’s audio information – even a CD likely sounds stunted to them (which would explain why our cats always seem to prefer analogue sources to digital ones)…

The masking effect (specifically the limits thereof) where the theory says lower amplitude sounds are masked by higher amplitude to the point that the lower are not in fact heard at all if the difference is large was brought home to me pretty forcefully with the new Orb album recently. When I first got it, it was a 192Kbps mp3 version which I listened to for a while before eventually going out and getting a cd version. When I got the cd I was fairly shocked at the difference – the latest album by the Orb has very much gone back to their original sound where the sound is layered and extremely complex, built up over the course of the song, and in many ways felt as much as physically heard – the mp3 had stripped so much of that out that it was almost like listening to 2 totally different albums…

@General Lighting 454438 wrote:

My source material is currently (unfortunately) mostly 320k MP3s with vinyl transfers being recorded at 44.1K/16 bit (as I think thats the standard of my USB turntable and also the mixing software I am using). This is being fed through an analogue mixer and into the Tascam line in.

The mp3’s are a waste of time recording over 44.1, due to the above mentioned upper cap on the highest frequency. Even when mp3’s are recorded with a 48K rate, the compression algorithm essentially removes most of the benefit of the higher frequency by focussing on the theoretical human audible response – essentially the fletcher/munson curve of between 2 and 5KHz and stripping out much of the data at the extremes in order to do this (even a 320Kbps mp3 is running at less than 1/2 the bandwidth required for 44.1KHz and 16bit – increase the frequency and more has to be stripped out – ditto the bit rate. Some of that compression is done losslessly, but not that much)…

The vinyl is a different matter, and would benefit from as high a rate as you can get away with – I tend to like recording at 96KHz and 24bit if I can, as the resulting recording sounds far better at that rate (and even taking into account the need to dither for changing bit depth, and anti aliasing for the sample rate change, the resample to cd quality still usually comes out with an acceptable sound)…

As for the DV, sony stuff tends to use their proprietary codec for the audio – either the pca lossless one, or the lossy one that I cannae remember the name of off hand… They are both (though it pains me to say anything good of the evil empire :cry:) pretty good quality, but are going to need downsampling (and therefore anti aliasing) to put on a cd (and will sound better at the 48KHz original rate if you can use it)….

[noname]

If you can find something solid to aim the direct dispersion from the speakers (ie what comes out of the front of the boxes) you will get much less leakage – we tend to use hills (of which there many in Scotland), and you mentioned it being a foresty/hilly area. The main thing is to get something solid between the speakers and any place where the leakage would cause a problem (the hay bales idea is a good one too, as hay bales would work in a similar way to the foam used to soundproof studios)…

Echos and reflections are much less of a problem, as the intensity of the sound is greatly reduced when it’s reflected so I wouldn’t worry too much about that – the important thing is to get something to interrupt the direct path to anyone who might complain.

Most of the time when people put sound equipment into valley’s, there is a tendency to point it along the line of the valley – never worked out why, as thats the easiest way to get noise complaints, but in my experience it’s what happens. If they set up at right angles – ie pointing into one of the hills, they would have far less trouble (as I say, this is how we always set up, and we never get noise complaints :love: )

[noname]

Ping statistics for 66.228.118.51:
Packets: Sent = 4, Received = 4, Lost = 0 (0% los
Approximate round trip times in milli-seconds:
Minimum = 168ms, Maximum = 169ms, Average = 168ms

Ping statistics for 67.228.118.21:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss
Approximate round trip times in milli-seconds:
Minimum = 199ms, Maximum = 200ms, Average = 199ms

Ping statistics for 208.43.118.51:
Packets: Sent = 4, Received = 4, Lost = 0 (0% los
Approximate round trip times in milli-seconds:
Minimum = 122ms, Maximum = 123ms, Average = 122ms

🙂

[noname]

Why don’t you just decompress it with the FLAC frontend (which you can d/l an installer for here) to wav and then work with it in wav…

[noname]

Get a roll of 20amp mains cable and wire them yourself – speakons are the simplest of all cables to make – no soldering necessary (unless you want to tin the ends), you just insert the cable and screw it down. You can get the plugs for £3.74 inc VAT and 100m reel of arctic grade 20amp cable for £89.12 inc VAT which is PVC insulated and sheathed, very flexible and will take temperatures down to -40C should you ever have a party in the arctic :wink:.

Will do you a good few cables, and will definetly carry any power your amp can put out…

[noname]

@DaftFader 387191 wrote:

yeah see the reason i ask this question is i have a roland groovebox mc-303 and i been trying to get it to work with midi how i discribed above but i don’t think it lets you do it – i was wondering if all hardware synths were the same … for instance would an emx or virus ti be the same?

An MC-303 will do it (I have one of those too). You just need to be sure and set it up properly and get it talking to cubase, then use MIDI learn to program it (you can do it by hand, but it takes a while). Roland essentially wrote the MIDI standard, so their boxes are usually the most faithful adherents to it…

[noname]

You can make a hardware synth do this, but you need to use the MIDI learn function quite a lot, and it takes a while to get it all working right…

But yeah – as long as the keyboard speaks MIDI (and pretty much all do nowadays) you can make cubase respond to it. As far as I remember you can also get downloadable “instrument” MIDI control templates for many hardware synths, but it’s a long time since I used cubase, so I couldn’t tell you how good they are…

[noname]

BBC News – Man arrested over illegal internet rave plan

Why using facebook and such to organise parties is (as has already been said) bait… :you_crazy

[noname]

@General Lighting 380586 wrote:

only problem is FB will turn over who did it, they will subsequently get busted for wasting police time..

That’d be a tough one to prove in court surely? They’d have to get evidence that the person posting it did it with the intention of misleading the police (which is likely to be a little difficult when all it really is is the police wasting their own time by believing what they read on t’intrawebs).

Might be a little difficult for you if you rang them with a tip off about a rave and they tracked you down – then maybe you’d get it for wasting OB hours, but don’t think any court would convict a case where the OB think you posted it deliberately up on some website you thought they would probably use to send them somewhere else. Even the most up tight judge would laugh that out of their court (and some of the more mental ones would likely do the OB concerned for contempt, or wasting court time or such)…

TBH the more I think about it, the better the idea seems – almost a genuine use for a farcebook. If we ever end up having trouble getting bacon free sites to party I may well consider doing it (only consider though, as actually doing it would be a terribly naughty thing to, and not something I would ever be involved in) :wink:.

[noname]

Don’t have a FB or a myspaz, bebo etc etc account, and never plan to get one… I’m a total social networking luddite (cept for PV and one or 2 other forums where my net footprint can be kept manageable)…

For me it’s a total lack of trust thing with most of the places – ever since the corporate web takeover started I’ve had problems with their whole reason for doing some of the things they do, and I am especially suspicious of any big corporation, or venture capital backed network that has no obvious profit stream (because that means they must have non-obvious profit streams)…

Mostly they claim non-invasive marketing research and targetted advertising as their profit (along with others like google, yahoo etc), but that to me sounds like a quick glossing over – for the market research targetting to be meaningful it needs to be at least a little invasive, as it needs to connect you with the things it is targetting you as maybe wanting (so at the very least some form of link tracking connected to your IP or username).

In the case of something like Amazon where they track what you look at on their site, and try and suggest things you might be interested in the profit mechanism is obvious – they can sell you more stuff (hopefully). In the case of google, the reason for the tracking metric is also relatively obvious (their whole search engine is based partially on what you do after getting search results from them), and the profit can be seen to come from the advertising they do based on your searches (although I am slightly suspicious of google these days as well, and how much info they collect and store).

But with Facebook I don’t see the profit aspect – considering it’s had over £300 million in venture capital you would expect to see an obvious way for it to make money (it claimed it’s first positive cash flow in 2009), but (and I may be being dumb here), I see no obvious way. Advertising is the only revenue stream I can see, and considering the amount of trouble google goes to to get ad cash, you would expect FB to be equally pushy ad wise, but it isn’t…

The fact that FB has the ability to collect FAR more info than something like google also worries me (plenty of people have very intimate facts about themselves published in their profiles), and their willingness to roll over and be the tart of any authority that asks them to makes putting any info on there a nono AFAIAC- at least google tell the DoJ to get bent when they get subpoena’d for the stored data – and fight and win their case. FB wouldn’t even fight the subpoena (as they say in their privacy statement).

The rest of the stuff like the censorship nonsense (I don’t really believe in censorship as a rule – most of the dumb stuff that ends up censored actually gains credibility by being censored – if exposed to the light of reason and debate, it generally withers and crawls off into a hole where it belongs), the various privacy screw ups like making loads of previously private stuff public without telling you, and the

Quote:

We may use information about you that we collect from other sources, including but not limited to newspapers and Internet sources such as blogs, instant messaging services and other users of Facebook, to supplement your profile.

thing just reinforce the cold feet really…

Hmmmm. Got carried away again (and a tiny bit off topic)… Still, my 2 penneth for what it’s worth :wink:.

Oh yeah . :rant::rant::rant:

And putting info about free parties on FB is so far beyond stupid it’s reached the calm waters of drooling gently into your chest that results from too much breakfast cereal…:you_crazy

If you’re doing it about parties you aren’t organising then you’re putting those that do at risk unnecessarily (getting people to free parties doesn’t need you to shout about them – people will get there anyway), and if you’re doing it about your own parties then expect to find trouble at every turn… Maybe not immediately, but it will happen (unless you are very very lucky)… :rant::rant::rant::rant::rant:

Using a FB group like that for disinformation purposes though :idea:….. That might have possibilities (post saying your party is one side of the county, and actually have it 100 miles away on the other side of the county…)

[noname]

@cheeseweasel 380333 wrote:

Scoops were traditionally what all the Jamaican dub sound systems used for bass. They do look fucking cool, but tend to sound sluggish, so aren’t great for music with fast bass like house and techno. Dnb and dubstep tend to have slow, halftime bass patterns so you can maybe get away with them if that’s all you play. I think the reason dnb/jungle systems often use scoops is a lot to do with the music’s dub roots.

If you’re going to be playing house on the system, personally I wouldn’t go with scoops.

The best subs I’ve heard have just been a box with 2×18″s in them, no fancy horns or ports, just really well-designed and with good quality drivers.

+1

These days we have drivers that can take serious high power, and the way to get the best sound out of them is a simple ported cab (ported because it means the box size doesn’t become ludicrously large). Scoops and horns add colouration by reinforcing a particular frequency in an unnatural way – was good for reggae rigs back when high power meant a few hundred watts, as the music benefited from that particular kind of reinforcement in the bass, but nowadays you’re better off designing a rig that can handle any music type, and then using an EQ to get the particular sound you want.

We have a pair of 2×18 ported cabs designed for the particular drivers we used (colossus xb), and they’ll make your teeth rattle. They take 1200-1800 watts each box, and will take a peak of about 5KW raaa.

[noname]

@MisterDuck 376853 wrote:

neither

LOL – ambisinister like me (incompetent with both hands)

[noname]

Pretty much what GL said – get a hardware comp/limiter and stick it somewhere inaccessible to anyone who isn’t meant to be twiddling with it – doing it with a PC is gonna leave it open to fiddling, and adds an unnecessary double conversion into the signal line (the PC will do an AD on the way in and a DA on the way out – unless you have a good soundcard on the PC which will come with a realtime comp/limiter anyways, you will get lots of unwanted nonsense in the signal that you really needn’t have there).

A behringer or a DBX will do the job perfectly well – a composer pro is only about £70-£80, and a dbx266 is somewhere close to that – both would do the job far better than a PC…

[noname]

Track on the B side of Bitch by Rotterdam termination source – thats pretty hardcore (cant remember the name of it the now, and I’d need to go to my attic to look)…

Think Lenny Dee gets my vote as the most hardcore person ever though – he was being a bammer when I was 15 years old (and thats now a fairly long time ago), and is still being a headcase now… Very hardcore person indeed raaa

[noname]

:laugh_at::laugh_at::laugh_at::laugh_at::laugh_at:

raaa

[noname]

That video is just wrong in soooo many ways…

Does anyone else find Ronald McfeckinDonald quite sinister? I mean more so than the general sinisterness of all clowns? Theres something very creepy about loudly dressed people with huge feet who wander around grinning all the time. Up to summat IMO :pee:

[noname]

Well…. That got all rather heated for no really very good reason :argue_mad :lol_crash.

Is it mebbe time to point you’s all in the direction of RFC 1855 again? (Not singling anyone out for this – most of the people who’ve had a say here could use reading it. 😉 )

The question asked at the beginning was perfectly valid, and expressing surprise at the subscription rules for advertising here is a fairly usual reaction if the reasons behind the subscription rules haven’t been explained (and no, saying the server costs money to run isn’t an explanation – it’s stating the blindingly obvious.)

As I remember from the discussions leading up to the change in rules, the main reason for asking for a donation to advertise pay events (or other money making ventures) was that having it free for all resulted in people quite frankly taking the piss – the sheer weight of advertising crap not only made the forum challenging to moderate effectively, but also caused the bandwidth/server costs to spiral. It was felt that people wanting to use PV for advertising – and there were lots of users that only came here to promote their businesses and contributed nothing else should be made to foot some of those spiralling server costs.

I would have thought a user who’s been registered a long time, and has made donations in the past as well as taken part in the discussions on the forum would have a pretty good case for getting something listed – did you PM Dr B Col? He’s usually pretty reasonable… The whole pay event/free party argument has to be at least slightly flexible – after all, if you put on free parties regularly the money to do so has to come from somewhere… Doing the odd commercial event is as valid a way as any of getting the cash needed together – in much the same way that asking people who advertise them here should subscribe (relying on donations exclusively regularly leaves you penniless after putting on a party, and the same is true of running high volume forums.) Nothing to do with the site becoming more corporate or whatever, just another way of keeping the free party spirit alive and kicking…

Hmmm – that turned into a rather longer waffle than intended… Hope it makes some sort of sense (been awake and partying for 3 days, so it could be just random dribbling, for which I apologise if that is the case :laugh_at:).

Remember :sign0015:… And that includes :sign0083:

[noname]

Been self employed all my working life – best piece of advice I can think of for getting clients etc is that the personal touch always works best. In your case that means doing some legwork, and knocking on peoples doors – even if you’re only handing them a flyer with the info on it you’re more likely to get their business if they’ve met you face to face.

And if you’re self employed, get yourself a good accountant or learn to do it properly yourself – you can put all sorts of things through your company books that you don’t expect (some friends who are bookdealers get their catfood and vet bills put through the company books – they are considered “rodent control staff” :weee:)… If your books show a profit in the first year, your accountant probably isn’t doing his job properly..

Good luck raaaraaaraaa

[noname]

Rig design

In this section we will look at what goes into designing a rig, and how to go about doing it. By the end of it you should hopefully have a fairly comprehensive design brief for the rig you want to build, and an idea of what it will likely cost to build. Obviously money is going to be one of the major factors in many of the decisions (unless you’re lucky enough to have unlimited funds), but there are many other factors that will make your life easier down the road if you think about them at the design stage…

Planning for a particular purpose.

The first thing you need to think about is what your rig will be used for – not in the sense of what music will be played (because a good rig design should be capable of reproducing anything well), but in the sense of how and what you will use it for. Will you be taking it outdoors using a tent and generator, or do you plan to only do indoor parties? Does it have to be easily portable, or are you building it into a venue? Will it be used for organised events or underground parties? Working out what you want to do with it now, and what you might want to do with it in future will save you a lot of time and effort later on because any need to modify for the new use will be already thought through and won’t necessitate rebuilding from scratch. To try and give you some idea of the sort of things you might take account of I’ll use the design brief for the rig we currently have (but definetly don’t do free parties with 😉 )

Quote:

Likely uses: Outdoor parties and large building parties (warehouses etc). Not too worried about small indoor events. Not building for hire, so an engineer will always be present to oversee it’s use. Set up and striking of equipment needs to be quick (for doing ninja parties, and the occasional need to exit a site sharpish.)

Things to take account of (in no particular order):

Outdoor venues require more acoustic power (due to the inverse square law), and need extra reinforcement in bass particularly.

Needs to be portable and quick to set up/take down. Must be able to achieve this with only 2 crew (becomes less important as we find new crew members, but keeping it achievable by as few as possible to be kept in mind).

Must fit into a single wheel base van (because twin wheel base and box vans etc require a group 7 MOT which is unneeded stress). So max packing size and weight roughly 12 cubic meters and 1350kg (size of a LWB high top transit) for all equipment (not just sound kit).

Modularity – needs to be easily reconfigurable (within reason) to take account of situation (ie if an amp failed, can the rest be easily tuned to take up the slack?) Needs to be easy to upgrade for when we have more cash to spend.

Sound – Must sound sweet (goes without saying really, but high audio quality central to the design). Keeping the sound intensity wanted for a party, but also keeping it contained so we’re not annoying people miles away (and therefore see the feds less often). High efficiency – we want to produce as much volume as we can with as little power as possible (while maintaining sound quality) – willy waving over how many watts your rig uses may impress your mates and punters, but only really shows your lack of skill as an engineer (how many engineers would be proud to say “I used 20KW to do something that can be done with 10”?)

Power requirements – outdoor rig, so generator powered. Low electrical demand important as lower power generators are less hassle to haul around the place. Preferably keep power single phase, as 3 phase genny’s are bulky and weigh lots. Multiple genny’s are a pain too, so whole system (including lighting) must be powered by no more than a 15KVA genny (so power max of 12KW, but the lower the better).

Robustness – It’ll be hauled around all the time, often to inhospitable places, and often set up when dark, so must be able to withstand the rigours of the road (and depredations of roadies) as much as possible. Easy servicing would be a bonus too and easy troubleshooting for when it stops working in the middle of a field. Looks not really important, as long as sound is good (so no need for carpeting speakers and such). Terrain likely to be rugged, so carrying handles rather than wheels (which will likely just get stuck), and individual boxes kept light enough to lift (no 30U racks with nowt but amplifiers).

Uses – predominantly DJ’s with turntables and DJ mixer, but provision for live acts desirable, so small professional mixer to be operated by the engineer a minimum requirement (mixer controls between DJ and rig desirable to avoid damage by over-excited turntable gorillas too).

That was the basic outline, and we tried to be as comprehensive as possible with what needed taking into account (grading by importance can be done later). Obviously you are probably going to miss things, and can’t plan for all eventualities you might encounter, but a solid idea of what your rig will be spending it’s time doing and some forethought given to any potential issues that may become important will save you a lot of stress in the long run.

Designing for modularity and robustness – why you should.

Irrespective of the use you are designing for, you should always put plenty of thought into both the modularity and robustness of the design – robustness means toughening the design as much as possible against everyday wear and tear and possible mishandling/misuse or accidents (all of which are not only possible, but likely to happen to a party rig). To that end it’s important to make sure everything is protected, so the design needs to include flight casing for electricals, grilles to protect speaker drivers and prevent pissed people trying to put their pint through any vent ports, minimal plugging/unplugging when setting up or taking down (the sockets on devices eventually fail from this – if you minimise it by hard wiring wherever possible in a rack you’ll extend the life of the equipment and save giving yourself some white hair tracing faults).
Modularity is also to be strived for as much as possible because there is no way to predict for a certainty what is going to happen in the future – making the design modular will help enormously if you find a new use you hadn’t considered at the start because it allows for some redesign by changing how the various modules connect, and allows for the insertion of new modules (as an example, in our design brief we originally dismissed the notion of small indoor venues, but later discovered that it would be useful. Owing to the modular design, we were able to reconfigure by adding a module and taking some out to allow this possibility).

Efficient use of time – what to design and build yourself, and what to buy ready made.

It is possible to build an entire system by gathering the components and building everything yourself – none of the electrical equipment is outrageously complicated in design, and even speaker drivers are constructable from their component parts if you are keen. For most of us though, it is useful to work out what has to be bought and what can be built – this doesn’t necessarily come down purely to a question of funds available, because it’s sometimes easier to achieve the design goals by building rather than buying even if the components used to build cost more than the ready built item would.

My advice on this is to buy everything pre-made apart from the speaker cabinets and the cabling. The speaker cabinets because that is the area of the overall system that is by far the least efficient, and large gains in overall efficiency can be had by devoting attention to designing and building them yourself. Cabling because it’s relatively easy to make, the components cost far less than pre-built, and making them yourself allows you to build custom cables for particular jobs (plus as long as you keep some components in your toolkit, you’ll never have to desperately hunt to try and find a replacement for the one you left on the kitchen table – you can just make another.) Flight casing is also relatively easy to build, and custom made cases can help make it modular, but I don’t tend to do this myself as pre-built cases cost about the same as the components required for building and are easy to modify to requirements (the one exception to this is the flightcase for our turntables which was custom built).

Design flowchart, and how to estimate build cost and time.

• 1. Arrange your list of things to take account of in order of importance, and split them into the categories Essential, Desirable, and Useful.
  2. List all the parts you already have that could be useful.
  3. Use list 1 to make design decisions – try to use the simplest method you can to solve the essentials first, then the desirable and finally the useful.
  4. Now list all the components necessary to those solutions, including the ones you already have. Don’t forget things like cabling, or wood, glue, screws etc if you intend to build speakers.
  5. Research all the possibilities you can come up with for the components, and ask opinions etc (places like PV, or the usenet group rec.audio.pro are useful for this).
  6. Find prices for the equipment (if buying second hand from ebay or the like, use the average auction sale price rather than the lowest you can find). Download and read the manuals for any equipment you consider so you know exactly what cabling will be required, and how many units of a rack it takes up etc.
  7. Add up the equipment cost (use zero for stuff you already have), and (most likely) gasp at the size of the number you arrive at.
  8. Now it’s time for the compromises – the aim is to reduce that huge number in 7 to the amount you actually have to spend – there may be things you can live without altogether, or cheaper alternatives to essential items that will do the job.
  9. Build time is more difficult – if it’s simply wiring the boxes together into flightcases it can be done in a matter of hours, and the time is going to depend mostly on how long it takes to acquire the component parts. If you are building speakers, allow up to a week for each pair. This should give a rough estimate of how long it will take.

Again, to attempt to make this process clearer I will use our real world example:

Quote:

1. Essential
Sound quality must be good.
Portable and able to fit single wheel base van.
Quick to erect/un-erect.
Sound intensity must be high enough for the great outdoors.
Mixer between DJ and amplification boxes for engineer use.
Modularity.
Robustness.

Desirable.
Easy upgrade potential.
Plenty of bass reinforcement for outdoors.
Power requirements within reach of single phase generator.

Useful.
No requirement for multiple generators.
Keeping required crew to a minimum.
Easy servicing and troubleshooting.

2. Equipment already owned:
Decks and DJ mixer in flightcase.
1 X Peavey M2600 power amp (260W RMS @ 4ohm)
2 X 18″ Black widow speaker drivers (400W RMS each).

3. Design decisions.
The first thing I noticed was that several of the requirements could be addressed by raising the efficiency as much as possible (portability, packed size, speed of erection, sound intensity, bass reinforcement and power requirements would all benefit from more efficiency.)
The second thing was that designing in modularity would help with the packing, upgrade potential, service and troubleshooting and numbers of crew needed.

The efficiency was a relatively easy problem, and only required the extra workload that comes with designing and building speakers. Making the whole system active (with respect to the crossover matrices) would further enhance the efficiency.

Designing speaker cabinets to each take only 1 driver type would help both efficiency and modularity over multiple driver type designs, and several small flightcases for amps, crossover etc would allow for some more modularity.

Using ported radiator designs for all cabinets other than highs would help with both the size issues and the set up speed – I’m less than convinced of the benefit of folded bass designs for anything other than large scale PA, where the longer throw is useful to cover the extra distance. In small scale (ie smaller than stadium size) a radiator produces more output into the nearfield than any horn design I’ve heard. Ported radiators also have the benefit of being smaller, easier to build and importantly much easier to set up than folded horns or scoop bins. The only place a horn flare is worthwhile IMO is with a high frequency compression driver where the exponential flare can cause the acoustic transformer effect and result in a large increase in volume. This benefit outweighs the extra time required to align them properly during set up.

Designing the mid range cabinets so they can effectively reproduce bass wouldn’t adversely affect their mid range performance, but would allow for more options in terms of modular configurations.

Designing and building the cabinets myself allows me to effectively over engineer all the joint fixings and material strength resulting in a much more robust speaker system.

Using a 3 way stereo crossover that could be run as a 4 way mono one would allow plenty of room for a future upgrade by way of getting another of the same crossover and using both in 4 way.

A pair of 10U racks would allow modular use by splitting up amps while at the same time ensuring the boxes don’t become too heavy to carry easily. External sockets on the racks would deal with necessary connections between them, and hard wiring speaker cables, input feed, and internal connections would allow them to pack into the cases and remove the need to constantly plug/unplug all connections.

Components
So from those solutions we knew our system needed to be 3 or more way active (preferably with upgrade potential via other modes), would require at least 3 amplifiers, and at least 2 10U racks. In addition we would require at least 3 speaker pairs of varying size, and enough wood to build the final speaker designs, and a mixer for the engineer. We already had a pair of 18″ drivers, decks, dj mixer and flightcase, and a 260W amplifier.

The component list minimum was:

2 X HF compression drivers with bolt on horn flare for each
2 X Mid range drivers (we decided on 12″ drivers which would be capable of bass reproduction if wanted).
2 X Bass drivers (Fane colossus would be preferred, but the black widows were already paid for)

4 sheets 8 X 4 marine ply (1/2″ thick)
1 giant bottle of glue
1 big box of 1″ screws
8 lengths 1″ X 1″ beading for internal strength
2 feet 4″ bore plastic pipe (for port tubes)
2 X 12″ speaker grilles
2 X 18″ speaker grilles

1 X 3 way crossover (with option for more ways in mono)
1 X Compressor for use as a level limiter to protect the amps/speakers

1 X amplifier to drive the compression drivers (the M2600 seemed perfect)
1 X mid amplifier
1 X Bass amplifier

2 X 10U flightcases (wood with metal edging/corners rather than ABS plastic)

1 X engineers mixer with flightcase

100m 3 core mains flex (min 16A power rating) for speaker cable and mains power inputs
100m good quality PVC sheathed mic/signal cable for all other cabling purposes
Connectors – at least 12 XLR female, 12 XLR male, 10 TRS 1/4″ jack, 8 speakon sockets, 4 XLR female sockets and 14 4 way speakon plugs

After my research I added my list of prices together, gasped (it came to over £6000, and I only had £1500), then sat down to work on those compromises. I ended up striking the compressor from the list as I found a crossover that had built in limiters (and was about 1/4 the price of my original choice). I picked up a good 2nd hand amp for the mid for £100, and got what I thought at the time was an excellent deal on a Yamaha CP2000 for the bass (£400). A pair of Eminence gamma 12’s, and a pair of 2″ DAS compression drivers with horn flares set me back another £200, the crossover £60, 2 10U flightcases cost me £140, and a second hand Soundcraft 400B in full flightcase was a steal at £190. That left me £400 for the speaker materials, cabling and connectors which was easily manageable.

So I’d sourced all the necessary components and managed not to require sacrificing any of the desirable or useful wishes on my list – it wasn’t my perfect system but it’d do the job. I estimated 3 weeks build time (and having had plenty of speaker building experience beforehand got it pretty much right). It sounded good, and was very efficient – under 2KW of power being used, and it was acceptable for outdoor use, could be set up in about 1/2 an hour and packed as quickly, and was solid enough to stand the rigours of road use.

The built in upgrade potential was useful too, because a year later a little carelessness resulted in one of the black widows and one side of the CP2000 being totally fried (I still have the CP2000, and will fix it as soon as I can source the parts). At the time I had some cash to spare (about £1200 or so), and decided to go 4 way as well as using my preferred bass drivers (18″ Fane colossus). I replaced the CP2000 with a pair of Peavey CS800’s running bridge mono, and built a pair of 15″ low mids using a pair of Fane 15B’s. Power use went up to about 3KW which was OK as it sounded like a rig several times that size. The current incarnation has been upgraded again (I got another pair of 18″ colossus and built the twin driver cabs I’d been choking to try out which have been every bit as good as I hoped they would be), and power use is now around the 4KW mark (with bass output heavy enough that the vibrations distort your eyesight anywhere within about 30 feet 😀 )

Your own list may be very different to the example here, and your solutions too. If you put plenty of thought into it though, and try to make your solutions cover or help as many of your list points as you can then you won’t go far wrong, and should end up with a useful design.

Power – working out the requirements, and working within safe limits.

Power is always going to be a limiting factor in any rig design – and it’s important to have some idea of how to calculate the requirements. The most power hungry devices are always going to be the amplifiers – they have a published power rating expressed as an average over a set time period (the RMS power rating). However, the power draw may be much more than that average, such that it becomes prudent to at least double the RMS rating when calculating power requirements – so rig using 4KW RMS amp power would require an 8KW supply – many people will tell you it’s possible to run on much less, and that’s true providing you don’t mind reducing the lifespan of your equipment and can live with the occasional blown driver or amp. If you’d like them to last I’d recommend powering them properly. You also need to account for any lighting, turntables, mixers processing etc etc in this calculation – most devices will have a power draw rating listed in their specs which you can add on to your doubled amp RMS value. I’ll go into this in more depth in the last part of this guide on rig power.

Matching amplifiers with speakers properly.

There are many opinions as regards matching amp to speaker power – some say one should be lower than the other, usually giving some quasi-scientific waffle as justification. The best advice and reasoning I have found is contained in one or two of Peavey’s tech notes, and specifically the “How much power” PDF document. Basically it boils down to matching the speakers program power handling value to an amps RMS output value (so an amp that outputs 400W RMS @ 8 ohm should be matched with a driver that is rated for 400W program power and 8 ohm.)

Under powering a speaker is probably slightly safer than over powering (after all, it’s more likely to drive beyond safe limits with an amp that is too powerful), but not by much because there is a tendency to try and drive underpowered drivers harder resulting in overdriving and distorting the signal (which is also very bad for a speaker)…

Linking it all together.

Once you have all your components, the next thing is to make a schematic drawing of the whole system, what connects to what, and which connector type is needed to make those connections. It will hopefully look something like this:

This can then be referred to if you are making your own cables to buy connectors, or if buying cables what type of cable you need for each connection, and can be used later on during the build process. Obviously your signal diagram will be specific to your set up, but you should try to keep it as clear, concise and uncluttered as possible – adding only necessary information (cable lengths would be useful additions for instance, whereas pin assignments for connectors can be noted elsewhere). If you’re hardwiring into flight cases, and using sockets to connect the cases together include it in the diagram. Another diagram detailing the power inputs and distribution within the cases would be useful, including peak power draw values and connector or distribution board types – allowing you to ensure the power system is robust enough to cope with the loads it’s carrying without spontaneously combusting…

Once all this is sorted, you’re ready to build (which will be the subject of the next section)…

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