SiWatts69

I reckon quadcabby has summed up pretty nicely there. The only things I'd add: Try to use the *exact* same mic for the same purpose each time. DI based inputs (keyboard, bass plus radio mics etc) typically don't need trimming each time, but wired mics (drum in particular) will. Using the same mic for the purpose each time will eliminate some of the variation. Contrary to what you might hope though, I've found that whilst little occurs when we re-trim, the "mix" still needs setting on a per venue basis. A perfect mix in one venue can sound muddy or shrill in another. The size of the room, number of speakers you're running out, subs and no subs all have an influence, not to mention environmental variables like soft furnishings etc all act to soak or reflect sound. Sometimes it will be as simple as playing with the master output EQ, but you're more likely to find that you need to adjust channel levels to find the sweet spot for the room. What you will notice tho is that by saving the setup (remember to save as a fresh setup each time you visit a new venue), you are already "nearly there" the next time. Each successive outing, you'll get quicker, not only in navigating round the UI, but in recognising the adjustments you need to make based on what you hear back. This is particularly easy if you save a recording. Just always take into account that when you soundcheck, record and then mix from the recording that the backline isn't present and adjust accordingly.

If all input channels are set to "linked" mode and their independent monitor gain levels haven't been altered then yes, what you'll get at the monitor output stage is a front of house mix (with or without FX) BUT step two is setting the actual monitor output level. Consider it to be the master fader for that monitor. Appendix D (D.9) On the PERFORM screen, select "Show Faders". Then at the top, touch the "Outs" button. You should then be presented with all the main output faders; monitor(s) and Mains (FOH) If necessary, increase the output. You can see the relative level on the fader meters. Do they look low, normal or high (ie are they running into the yellow, or staying well below 0 in the green? If the faders are already quite a way up, it suggests that your gain structure further down in the chain is out, eg trim levels set too high.

You don't say what the monitor speakers are or how you have them connected (L6Link or analog). There are three things governing the monitor volume(s) The amount of any individual channel fed into the monitor channel The overall output level for the monitor channel The gain setting on the monitor speaker. Check all three. Likely, you've some disparity somewhere in the gain structures.

There's nothing wrong with the power of the gear... it'll go mental loud! BUT, to give good control over the volume, the sweep is much more gradual. Don't try to compare how your guitar amp volume control works compared with how the L2 works. The L2 has to be a lot more versatile, and as such its gain controls work differently. I promise you, you'll still be able to generate plenty of volume, just likely in a more controlled manner.

The gain sweep on the side panel of your L2 is designed to accept both line and mic level signals. As such the apparent range of sweep for a mic appears (at first) to be quite high up, typically just beyond the 12 o-clock position on the side panel gain control. This is normal. Check that the pad switch is set correctly (0 for microphones, -20 for line level inputs like guitar/k/b etc)

There are physical differences between standard mic cable and AES/EBU DMX cable. Mic cables are a twisted pair shielded by a multi braided copper wrap. AES/EBU and DMX cables are a twisted sheathed pair with a twisted unsheathed drain wire, all wrapped in a foil shield AES/EBU can be used for analogue mic signals without issue and is appropriate (even if not perfect) for use for DMX DMX can be used for analogue mic signals without issue and is appropriate (even if not perfect) for use for AES/EBU Mic cable should be avoided for use with digital signals (AES/EBU and DMX) if at all possible although it will work along short runs. Where issues do arise, AES/EBU is the most likely affected. DMX is considerably more happy with the use of standard mic cable. BUT, the BEST COURSE OF ACTION is to ALWAYS TRY TO SOURCE THE CORRECT CABLE TYPE FOR THE APPLICATION. Anything else is simply adding to the risk. As I said first time round, most "DMX" cable sold in Europe is actually 110ohm, ie AES/EBU anyway. Any cable sold as AES/EBU + DMX will likely be 110ohm DMX runs should be "terminated" with a 120ohm bridge across pins 2 and 3 (typically by means of an XLRm plug with the resistor solder inside, plugged into the DMX out of the final light fixture in the run.

If it helps... Taking an SM58 as an example, for either me or our lead singer, auto trim typically sets the trim level at about 27dB With a Beta58 it drops in at around 23dB (3-4dB less as the Beta58 has a higher signal level than an SM58) We used to have a female singer using a Beta58 and due to her voice being somewhat weaker and her mic technique auto-trim typically set her Beta58 at around the 28dB mark. Trim is the amount of gain being applied to the input to get it to a nominal unity level. Auto-trim is trying to detect the highest peak in the signal and set a gain that is appropriate to handle those peaks without clipping occurring for anything other than quick peaks. At -30db on your masters, you are effectively amplifying excessively at the start of the chain and then taking off that amplification at the end of the chain. Ideally, you'd have trim set according to auto-trim (or close to it), then adjust the lead vocals to be around the unity mark (leaving a fair bit of headroom for in-gig adjustment) and set the other channel levels to get the mix from your lead vox channel. Then, bring up your masters to set an appropriate volume for the room. I can't remember ever running my masters as low as -30, having them typically sitting somewhere between -10 and 0 Remember, your mixer is only there to "shape" the sound. When using auto trim, each performer needs to perform at performance level for about 20 seconds and then you need to accept the changes suggested for it to kick in. We tend to auto trim the drum kit mic by mic. Vocals we tend to do as a group. Bass and my sax and (when we had them) keys typically don't need adjusting much from one gig to the next as they're all DI style. So; TRIM is an amount of gain applied to the channel to bring the signal level/strength roughly to unity 0db and thus into line with all other inputs Channel Fade is the relative amount of that channel's level fed to the main output mix Master fade is the summed level of all channels Too much trim will increase the risk of feedback.

The M20 will auto assign outputs to the monitors based upon their order in the L6Link chain... The first L series speaker in the chain "on its side", ie being used as a monitor, will be auto-assigned Monitor channel A, the second mon B etc. It is most likely that your M20 hasn't detected the L"'s as monitors and as such is feeding them either an L or R (or L+R) output. If you have the speakers upright, they will not be recognised as monitors. Check your L6Link settings. "Ping" the monitors to confirm which is which and the M20 will show you which channel has been assigned. If one isn't being given the channel you want, deselect the Auto Assign option, then manually assign the correct channel to the correct speakers, A, B, C, D. If you are running your L2's upright, you will need to do this regardless.

Without wishing to make what appear to be dumb suggestions: The main gain knob on the back panel of the L2t needs to be set to (typically) the 12 o'clock position (there's a notch) Then, bring the volume of the mic up using the "Gain" knob on the side panel control. Check that the stereo link switch is set to DUAL MONO, else only the right hand gain knob will do anything. To my knowledge, no phantom power is available on any of the inputs, so you should use a dynamic microphone. Condenser mics will not work without either an outboard pre-amp capable of supplying the P48 phantom power OR a standalone phantom power supply.

Glad you got it sorted :-)

The DMX protocol specifies 120ohm cable whereas AES/EBU is 110ohm. BUT many DMX cables sold are actually 110Ohm. Certainly, in Europe, almost all DMX cable is 110ohm and so is pretty much interchangeable with AES/EBU So, my suggestion would be to check the actual spec of the cable you've found. If it's 110ohm, it is actually more likely AES/EBU being sold as DMX rather than the other way round.

I managed to do this a few weeks ago... or rather, the person I'd left in charge of control hadn't stopped the recording and I flipped the power off. 2 options: I use mixcraft 6 as my DAW. I was able to drag and drop the files direct off the SD card and mixcraft imported them without issue, ie all audio was present. It is possible that other DAW software might be able to import the files, despite the header data being missing. This option is great if all you want to do is to import the tracks into a DAW. BUT the above didn't allow me to play back the recordings as standalone files. A quick bit of research on this forum pointed me to a small piece of software called "Format Factory" http://www.pcfreetime.com/ Use it to export the files as WAV (to a different location) and the missing header data is set into the new versions of the files. Worked a treat, and I was then able to put the copies back onto the SD card and play them back.

As with litesnsirens, I'd be wondering why the FOH guy would solo a channel mid performance too. But your point is still valid. On the basis that the M20d would be stageside and the FOH guy working from an iPad, he'd have no relevant need to solo a channel.... it would be of no use on the iPad If the M20 were not stageside, the option of running a headphone output would be moot anyway, since the run would (probably) be too far to make it usable. Also, I'd have thought any decent FOH guy would, by the time you are performing, be adjusting any eq in relation to what they are hearing through the main speakers, rather than in cans. BUT, using the headphone outs in this manner could throw a curveball IF a channel was left solo'd. Personally, we only solo a channel during a preliminary mix from recording where the preliminary mix is being done in cans prior to raising the main outs to tweak it for the room.

litesnsirens gives a damn fine idea there, assuming you simply want the FOH signal in your in ears :-)

Two options: First, yes, you can send the main mix to your in ears simply by connecting the Main L/R outputs to your in ear transmitter. If you need the main (analog) outs to route on to your amps, you'll need a stereo in ear system with pass-thru so the signals can continue on to your amps OR a 2 into 4 signal splitter (2 inputs to 4 outputs). Alternatively, the simpler method and as per quadcabby's suggestion, in *your* monitor channel on the M20d, set all the channels to "linked" mode and set the levels for each such that the indicator triangle is green. When the indicator is green, there is no boost or cut on the channel relative to the main out level for that channel, giving you a mono summed main outs mix. If you do need to boost your own (for example) channel, simply adjust the level of your keys into your monitor channel. The best thing about "linked" mode is that the input will follow the main out level set for it, so if you need to (for example) increase the main vox in your FOH mix, that change will be mirrored in any monitor channel where the input is present and linked. Perhaps your second guitar player needs to learn a little stage etiquette! All that "a lot of power" on the stage does is to create an inbalance in the main FOH mix (too much stage noise bleeding out) and the ongoing battle of rising stage volumes as individuals struggle to hear themselves and turn up "a little" progressively through the night.

Here's a live recording, again done on the M20d of a recent outing with a SKA band that keep hiring me and my rig... No mix afterwards, just captured the main outs into DAW from the SD recordings with the mix left exactly as it was for the gig. Sound on the night was awesome. https://soundcloud.com/handsome-dan-mavericks/beer-reel-big-fish-cover-live-2015

Take TheRealZap's word for it... !!! I'll get mi coat ;-)

A generation 2 will be fine. I can't see any reason why a 1st generation wouldn't but the few quid saved now might come back and haunt you in 12 or so months time.

A wise decision me thinks :-)

Before you jump the gun and assume that there is nothing extra going off in the dpa adapter, get it checked to ensure there isn't an embedded resistor in there. I'd hate for you to rip the microdot connector off an expensive mic, solder on a straight 1/4" TRS jack and then find that it no longer works. I made that imistake early on... I assumed it was just wire going in to a connector, made up a new lead, went to rehearse and the bloomin thing didn't work... and that was the AKG supplied cable that had wierd things going off in it (pins 2&3 shorted and pin 1 shorted to the sleeve. Run a multimeter across the in/outs of the dpa adapter and just check the connections have 0 ohms between the poles/pins.

I noticed the main outs mix is quite queiet too, even when blasting it out the main speakers! I've recently started playing back the full gig recording (raw input recordings) via the M20d and capturing them to a fresh stereo input track in my DAW by running it through a 2 channel audio interface (Roland Duo-Capture EX). I think Arne gave an explanation about why the master outs are at a lower than expected level. I'd not be suggesting doing anything with your input trims UNLESS they are set incorrectly for the mics being put through the M20d. Run auto trim on them with someone talking at te usual conference levels, then turn OFF trim tracking.

If the limiter is kicking in, you are over driving the speaker, most commonly through poor gain settings. Where do you run the master gain knob on the back panel? Line6 speakers are optimised to be run at the 12 o'clock (unity) position and running above this level could be your issue. There's a handy "notch" at the 12 o'clock position.

Love it António We've done similar in the past but no way as methodically as that. The section about taming errant egos is particularly apt.

Without taking input impedance into account, you'd connect the tip and ring together from the TBP06 onto the tip of the DPA connector/adapter. BUT there is no guarantee you will get a sufficiently high signal from the mic. You're using a microdot to sennheiser adapter, designed to match the DPA mic to the input impedance of a sennheiser transmitter. There is no guarantee that the input impedances match Your Line6 TBP06 has an input impedance of 1.2MOhm (though a TBP12 has an option to lower it to 20KOhm). A sennheiser G3 transmitter (for which the DPA adapter is designed) has an input impedance of; 40KOhm or 1MOhm in unbalanced. So, what I summise from the specs is that the DPA is designed for the 40KOhm impedance. You could try a simple convertor (3/8" TRS to 1/4" TRS) but if your signal is weak, you'll need to work out how to adjust the input impedance to make it match that of the TBP06. First test then is to create the adapter and see whether you get any joy.

Sorry, I understood the TBP06 to have a TS connector (1/4" unbalanced) input and not a TRS. However this excerpt from a PDF I just found appears to contradict that. If the TBP06 is supplying a bias voltage at the ring, then you're part way there. The next thing to check would be the expected load impedance (input impedance of the transmitter). I run AKG C519's into my TBP12 and I had to modify the load impedance by placing a resistor into the connector for the AKG mics to work. Without the load impedance being set correctly, the output signal from the mic, whilst there, was extremely low. You'll need to check the input impedance value for the sennheiser transmitter (that the selected adapter is designed for). You'll need to discover the input impedance for the TBP06 (Don might know this but what I've found is 1.2MOhm) You'll then finally need to know how to adjust any imbalance between the two values WITHOUT degrading the mic signal. Contrary to what you might think, there is possibly an embedded passive component (resistor) inside the DPA convertor that matches the mic to the sennheiser transmitter. Simply buying an adapter with the correct in/out isn't necessarily enouh. I did this for anyone wanting to use AKG micromics with their TBP12... http://line6.com/support/topic/12359-akg-c519ml-mic-with-xd-v75-and-tbp12-beltpack/