JBL SRX718s Crossover

jkeefauver

Just bought 4 SRX718 subs for my FOH. Tops are Mackie 1530's (active). QSC PLX2502's on the subs (bridge, 4 ohm, 1 amp per 2 cabs per side).

What do you guys like as a crossover point for these beasts. I need to set gain structure for all this but want to start with a believable x-over point. We were using around 90-100HZ on our old subs (too embarrassed to mention - starts with a P) but I assume it will be lower for this setup.

Thoughts?

Gadget

The sub doesn't set the crossover point, unless it's a horn loaded one that can only go to somewhere below 100 hz or so...There's more to the story here...The tops, and their power, and makeup are much more important to the crossover point than the sub. Subs can start to sound honky as they approach 120 hz, so we typically like to cross below this point...unless the top box is 10\" or 12\" with a 3db down point somewhere above that...there is no such limit here, but there may be some experimentation necessary. The 1530's start to fall off sharply @ 60 hz. That is probably the lowest point you would want to use for a couple of reasons. One (I tried to find a crossover for the midrange, but couldn't, but it looks like it's around 700-800hz...that means that whatever hipass we set, the lo mid on the 1530 will have to go from there all the way to 700-800 hz... the more the cone is expected to do in the LF range, the more it will move, end the greater the distortion potential in the lo mid sound. The subs have no such problem since they are not expected to do any frequencies where intelligibility is concerned. So, lets try and leave the real air moving stuff to the sub and that frees the lo mid up to reproduce the important lo mid frequencies.

If we look at what JBL has done, they have crossed the SRX715 to the SRX728 @ 90hz, and the SRX725 @ 93.4 hz. Granted these are not Mackie 1530 tops but it tells us something don't you think? The spec sheet suggests 80 hz, but that is a generalization, and I think a marketing ploy...The sub is most efficient @ about 120 hz, and from that point about 5 db down @ 40 hz. If we take the average of the response we could interpret the -3db they list @ 33 hz.. but that's pushing it....

The 1530's have a peak below 90 hz we can eliminate by staying above that frequency so lets try the 90-100 hz region first, then push it up to 110 hz , and 120 hz and see what we find. It's pretty darn easy to do, and if you want to get the most out of your system it will be worth it. I would then recommend:

38~hz BW 18 HPF
90~hz LR24 LPF
to start.. I figure your going to end up @ about 8-10 ms of delay on the tops depending on where you place them on the subs. The FAQ section has a lot of good info on alignment and optimization so go there.
Gadget

jkeefauver

Give me a couple hours to play with it and I'll let you know how it worked. Thanks G - as always.

Gadget

I spec'd 38 hz because of Power available, You are in the middle of the power range, so you could get a little louder with the higher hipass, If you want lower LF extension and don't need the volume (read that amps clip/exhaust faster) set the HPF @ 33 hz BW 18.
Gadget
P.S. I forgot to put the LPF in for the JL SRX715/718 combo.. I added it.

jkeefauver

Well, I'd say that the JBL subs were a HUGE upgrade. System sounds amazing. Have to be careful not to shake the pictures off the shelf inside my house 'cause I have this setup in my garage at the moment! Incidentally, left filter at 38 (like that better) and raised x-over on both tops and bottoms to 106Hz. Nice round sound, lots of punch and clarity.

Anyhow - thanks for the input Gadget. Maybe you can help me with one more thing...

Am training a new , beginner soundman and want a good way to explain gain, trim, channel level vs. subgroup level vs. master level to him and how he should structure things during shows. There seem to be about 10 schools of thought on this and I'm not sure any of them are right or wrong but I'd love to hear your take.

I can show him I think but I'd really like to be able to explain things a little bit as to why you do things a certain way. I learned by trial and error like a lot of people and I want to have a good explanation for him. I know this isn't necessarily a DR issue but you guys are a wealth of knowledge and I was hoping to pass some of it along.

FYI - board is a Mackie TT24 Digital, we're using the front 24 channels and 4 sub groups (vox, keys, guitars, and rhythm) for this setup. Also using 6 auxes fed pre-fader to IEM's. If this conversation isn't appropriate for this board, I understand, perhaps we can take it offline. Thanks.

Gadget

A quick perusal of the FAQ section should show you that we are about education and betterment of live sound here.. it's NOT all things Driverack.. that's my choice... and I'm the second tallest horse at the trough here LOL...I'm full of myself aren't I...

Well I don't know that board at all, but if it has PFL listen in the channel strip menu then the procedure is quite strait forward, and is basically covered in the welcome thread, and in Phillip Graham's more comprehensive work which is posted here:
http://www.dbxpro.com/forum/viewtopic.php?t=1475
Here is a caveat from that work...

A line check's primary purpose, to my mind, is to insure that all relevant signal inputs make it to the signal outputs. Any more than this is going to take too much time, resulting in mistakes, missed channels, etc.

A line check is generally not a time to fiddle with eq, adjust attack or release times on compressors, and other such diversions. If you can adjust such parameters quickly from memory on consistent sources, perhaps consider adjusting those parameters on one or two key channels (e.g. lead vocals).

I feel that a line check is essential, whether there was a soundcheck or not. In the case of no soundcheck, it is an essential component of making noise come out of the pa. In the case of a full sound check, it is a confidence booster that everything is still plugged in and working as before, and that no one has toyed with the console. In the festival situation, the system tech/babysitter may offer to help you with this, and I suggest taking advantage of that resource.

In my methodology for a line check I try to split tasks between what requires active signal and what doesn't. For instance, setting the gain trim of a channel requires the channel to have signal passing through it, but checking the bus routing of the channel does not require signal. Checking whether a compressor is bypassed or not does not require signal, but checking whether the insert point and compressor are passing audio does require signal.

If there was a substantial soundcheck, simply looking at the channel input meters (or cueing the channels on boards without per channel metering) while the band/tech/presenter are warming up and plugging in is sufficient. On most better boards, a quick scan across the channels trim is all that is required.
If the board is an analog board, i will then scan up/down the channel strips to check that inserts are turned on/off, eq is turned on/off, and that the routing (LR, groups, VCAs, etc.) is correct. On boards with detailed VCA assignment, or digital boards, this obviously is done in a different way. The essence is to check for signal first, then check for routing.

If one has the luxury of moving input by input across the stage, and doing a more thorough scan of the each channel while a tech/musician provides a source, then count yourself in a good situation. I personally don't feel I can depend on this happening at line check in every situation, but it is preferable
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Now, I like to put my headphones in and have all the channels muted as Phillip says and on a per channel basis either while the muso's are tuning up or running through some songs or get them to actually play something for my benefit should I have a problem getting what I need, or when there is trouble with a given channel.

Engage the PFL and listen to that individual channel and note the channels meter, and in the case of a mixer without individual channel meters (and often those that do, because of the greater resolution on the main output meters) use the mixers main output meters (because on any quality mixer engaging the PFL will route the sound to the phones (sometimes a little button pressing is necessary to route the PFL mix to the headphones...) and the meter will show the signal coming into that channel. Here we can perform a two fold purpose...LISTEN to the channel and see if any gross EQ is needed to get a round full sound on that channel, (and thanks to our properly tuned set up and flat system the headphone sound translates well to the mix...) and use the input gain control on that channel to get a good signal to noise ratio...On a digital mixer it's a little more important to stay out of the clip range (read that NO CLIPPING EVER) :shock: So it's better to have an average that never comes close to clipping on a channel with lots of dynamic's like Kick and snare. On an analog mixer we have signal to noise issue's to deal with so we want the maximum signal we can get to lower the noise floor. In our gain structure we learned that on an analog mixer, the design centers on the channel strips main faders...that little =0= thing about 3/4 of the way up where it goes from negative numbers to positive ones...we want the most important things, vocals, to be at that point when we mix, the other lesser components can therefor be pulled down slightly, or more depending on their importance to the mix. This way, we can look down and see a pattern and get a visual reference on the various elements of the mix...and where they lie within the sonic palate.

Ok back to the input gain control... We have the PFL engaged, done any gross channel EQ necessary (does GIGO trigger any memory engrams here.. usually a good sounding source , on our wonderfully tuned system will require VERY little EQ...A discussion of channel EQ is beyond the scope of this post.. maybe we'll explore that next) engage the HPF on any channels that don't have any business in the subs..I also like to lower the lo shelf on vocal channels to avoid a muddy mix. As Phillip said it's important here to also make sure that any inserted devices in the channel strip are passing sound and not clipping within themselves...As we watch the meters and listen to the sound, use the input gain control to adjust the level on the meters so that on a digital console there is at least 8-10 db of headroom (where possible) and no less than 4-6db...( on kick and snare you need to have them played as loud as they will normally be played) on an analog channel average at least -0- vu with occasional peaks(red clip LED) and I mean OCCASIONAL used as a tool to get that PHAT analog sound..( this is the natural result in the analog preamp as it compresses as the maximum gain of the amp is approached) is acceptable, but best in most cases to keep it so you have a few db of headroom on any given channel. I like to use the master faders to set the overall system volume, once the vocal channels have been set using the above method.. remember that this all assumes a properly tuned, system and a gain structure prior to this point... the reason? Far more noise will be induced with the multiple channels if they are not all set to best gain structure, than master faders @ the design centers.. AND we have less noise on the cheaper consoles with lower gain here. It also allows you to grow the volume of the mix as the crowd size increases...

So we have the input gain set, any outboard inserted devices passing sound and not clipping, and any gross EQ done before we ever turn the channel strip back on, disengage the the PFL. Now move on to the next channel...I can generally do this (on any given channel) in about a minute or less, 2.. on problem channel, and that's important when 24-32 channels are used regularly...although some are efx returns.. and 'ME' mic and tunes...and as such rarely require lots of attention at the setup stage...Those of you with group mutes and VCA's can set up those now, or those who have 4 or 8 buses can now engage a sub master routing scheme for \"GAS\" controls if you wish, I route 1/2 to aux master 1/2 for VOX, 3/4 for instruments 5/6 for drums, and 7/8 for lighting/efx or any other purpose like recording... where VCA's are absent it allows a global gain change for that group usually pretty subtle though compared with VCA's.. oh and the L/R must be engaged on those subs for them to affect the mix...

That's my slant on the subject.. any thoughts?
gadget

jkeefauver

Excellent - I think this will give me a good fundamental primer on \"why\" we do this stuff and how it interrelates. That board is quite full-featured with 12 auxes, on-board DSP and an extensive matrix section. You can also make the subgroups VCA's but I haven't really come up with a practical use for that just yet. I probably need to learn more about that as this is the first board I've owned that had them. I like the board a lot - not so much Mackie support - but the board is good. Anyhow, I will continue in my teachings with this guy and urge him to read this forum any chance he gets. I think he's tired of my preaching \"gain structure\" all the time but he'll learn!

Thanks again Gary.