Rick_Auricchio

I agree. I may not bother with scenes either; right now, I have one scene per setup. Besides, with only 32 scenes per setup, you really can't consider a separate scene per song---that's not enough for most gigs. One answer would be to categorize scenes: e.g. ballad, rocker, more reverb here or there, etc. That's way more mental trouble than it's worth when I'm busy playing and singing. At the most, I can see having four scenes with different vocal presets, since all four of us in the band sing lead at times. The first, with me as Lead Vocal and the other mikes with EQ/Comp optimized for backup vocals. Another for my keyboardist, third for guitarist singing lead, fourth for the drummer. This doesn't take much brainpower to flip among the four scenes. (As long as I never screw up!) (I do understand that changing Setups mutes all outputs for a second or two.) For me, understanding the architecture of a system---how things work---is so important. If I can think in the abstract about a system, I can easily understand the operational details. I'm never been comfortable doing something by rote, like "just save the scene and the setup blindly whenever you make a change." Sure, that works, but I don't learn or understand what I'm doing.

Is this correct? Took me several readings of the manual, watching the video, reading other posts, and several experiments. A SETUP contains the hardware config of the mixer, for each input and output channel. I/O connector assignment, trim, DSP preset chosen, monitor sends, FX sends, mute/solo, pan, level. (Not all of these apply to outputs.) Also the stage appearance (i.e. curtain etc.) A SCENE contains (for each input and output): stage icon, monitor sends, FX sends, FX parameters, mute/solo, pan, level, and all the Tweak settings applied to the existing DSP preset. A question: Is all that stuff really saved in the Setup, or is it actually contained in the first default scene that's automatically created when you save a Setup? This would seem to make more sense; at least it does to me. So if one changes a DSP preset to, for example, get a different canned EQ curve, this constitutes a change of DSP preset. So it will be saved in the setup, not in a scene. It took a bit of experimentation to arrive at the above conclusion. I wanted two situations: one with flat EQ on main output, the other set for a boomy room. If I chose the Boomy Room preset for the output, that wasn't saving in the Scene, but it was in the Setup. Then I figured out that if I manually tweaked a flat EQ to match that in the Boomy Room preset---without ever selecting that preset---that saved in the Scene. I realize this stuff has been stated several ways in the past, but it's going to confuse people. I'm not dumb: I've been a career software developer, owned a studio, used digital mixers, and it still was really difficult to get my head around this. Luckily I've been experimenting with the M20d for a few weeks---before ever trying it out with my band!

I just ran a test, and jmakela is indeed correct. The phones feed taps the main LR output before DSP and fader. The block diagram is incorrect (or the firmware has a bug). Either way, the phones will only pick up the main mix as if it were a prefader solo of the main mix. As for channel solo, the manual doesn't specify if it is pre- or post-fader; the block diagram indicates post-fader. But now I have my doubts. (I didn't test channel solo.) The spec sheet doesn't even mention solo. It should list pre- or post-fader solo.

I have not tested this. But the block diagram clearly shows that the headphone feed is tapped from the mains after the main DSP and fader. As pedromleite says, there is a level control for the headphones themselves, but any change to the main fader should still be heard. If that is not how it works, then the block diagram is wrong. Can someone from Line6 verify this?

If nothing is Soloed, the headphones should provide the exact same signal that's coming out the LR mix, with FX, master fader and all. Check that you don't have something soloed.

The block diagram shows the record/playback tapped from the input signal right after the input-gain section. So if that's correct, you should be able to simply pull the channel's fader all the way down to keep that signal out of the LR mix.

An additional advantage of the Airport Express is that it can be set to receive Airplay music. Just connect a 1/8" cable from the Express to the aux input of the M20d and you have music playback from your iPad or iPhone. One could even buy a length of dual-Ethernet cable (called "siamese ethernet") to keep the wiring cleaner. Install Ethernet connectors on half of the cable and the 1/8" plugs on the other half. http://www.hometech.com/hts/products/phone_data/bulk_cable/ht-cat6s.html

Wow, I'm certainly out of touch there. (I was always working in the OS and never in applications, so I suppose I have an excuse for not knowing about solutions like this.) Of course, one must learn C#; I'm great at C and good with C++, not that I code any more. For the past 12 years I've done independent Mac support; I might only code a little personal project now and then.

The manual says 17/18 will show as Aux In, so that seems normal.

That works too! I noticed in the block diagram that the headphone jack mirrors the main output if nothing is soloed.

It should work. From the user guide, page 4-8 where USB streaming is described: "Note: In order to hear the stereo return, a Stage Icon Preset must be assigned to the Aux Input using the M20d touchscreen UI. The Laptop preset is ideally suited for this purpose, and can be found in the Stage Icon Gallery." It seems you've done that. I'd check all assignments and faders again...even walk away for a while and come back to it.

The L2T has multiple audio-input jacks at various levels (line or instrument/mic). The output of that tube pedal is probably at instrument level so that it can drive a conventional guitar amplifier. Audio is audio. Just plug it in and go. As for whether you'd have a "great tube-ish sound," that's purely subjective. Your impressions and mine may differ. You have to listen to decide if you like it. (If this were not the case, there wouldn't be 500 different distortion pedals on the market!)

Neat. Those StageSource speakers have a few nice tricks up their sleeves. (Do they actually have sleeves?)

You can't route the main mix to an aux (monitor) output. The solution suggested by eklynx will work, because as you adjust each channel the linked aux send will follow. But any change to the main fader won't change the overall output level of the aux; you'll have to do that separately. There are tricks you can do with external patches; I've done things like this in the past. For example, split the main LR outs with Y-cables. Run those to your FOH amplifier and also to two spare Line input channels. Route those channels only to the aux you want, UNlinked. Do not route them to the mains at all otherwise you'll a have feedback loop. (This is a little tricky because you have to guarantee you never raise the fader on those channels.) Set the appropriate aux-send level in those channels. Now when you adjust the main mix (and its level), you get a corresponding level change in the aux output.

Well, these are only our opinions. (But as a 30-year veteran software developer, I think I have a good perspective on software.) Of course using IOS is an advantage to Line6, more than it is for customers---if it were harder to develop an app, they might never have done so, or it might have taken much longer to accomplish. Remember, any company has to justify the budget for developing an app. It isn't just software: there's a lot of group effort that goes into the design and usability testing for an app. As for an iPhone app, its existence is made possible because the app is developed under IOS. There's no technical reason the iPhone wouldn't work. (Of course, an iPhone app would need a totally different user interface due to the small screen size, but the majority of the code wouldn't need rewriting.) Is this true for Android tablets and phones? Maybe; I don't know. Are all models of Android tablets the same? Same programming interfaces, screen size/resolution, switches, buttons, etc? Again, I don't know. But if they're different, then this becomes a big problem for the app. You eventually end up saying "The app runs on these models of Android tablets, and no others. Sorry." Every vendor wants to standardize his product, so they're all alike for consumer appeal. Then each one changes his to make it "a little better" than the next guy. The result is minor differences across the product lines. It's not a problem for a consumer who only owns one device, but a nightmare for a software developer to write specialized code and interface elements for different models. A nontechnical reason I can think of is the availability of tablets. You can buy an Apple iPad most anywhere in the world; if you were to need specific Android tablet models, would this be the case? Again, it's opinion. I'm happy to have an app available. If Line6 went only with Android and it took a year longer to get an app, would you be happy waiting? And if your supported model broke and was discontinued, now you're shopping around?

I agree that iPads cost more than the plethora of Android tablets, but they provide several advantages for the development of an app by a company: 1. A well-documented software development system from one vendor, Apple. This includes simulators for testing on Mac computers before even working with the IOS device; 2. Software compatibility among IOS devices: most iPad and iPhone models; 3. Thousands of IOS app developers available to do the work; 4. Consistent hardware and OS, whereas every make/model of Android tablet could be different. These three factors make it much easier for app development.

Version 1.2 was released a year ago.

A minor UI issue in Setup mode, regarding the Stage Icon Gallery. IF you mark an instrument icon as a favorite AND that icon happens to be at the left edge of the current Gallery page THEN the left-arrow indicating a prior Gallery page is obscured by the star. This confused me for several minutes; I thought the gallery lost a whole bunch of instruments. I hadn't realized there were more pages to the left of the current page.

Just a bump to advise readers of the edits to the original posting.

1. Add a monitor-send "PRE" pickoff point at a point before EQ, compression, and delay. Having the pickoff happen after EQ/Comp/Delay makes it difficult or impossible to enable this processing in the channel, because that stuff will end up in monitors. Compression is especially useful but it can contribute to monitor feedback. Many singers don't want delay in the monitors. An ideal choice would be three pickoff points: 1. NEW: after FBS, HPF, and Gate but before any EQ, Comp or Delay; 2. EXISTING: pre-fader but post-DSP 3. EXISTING: post-fader 2. When deep-tweaking a channel, the monitors should show as the names given to them in the setup, rather than the generic "Monitor A" etc. 3. Add a setting to prevent Setup mode from looking at input connections when a setup is loaded. If I want to make a change to a live setup in my home studio---and I happen to have some extra stuff plugged in---the setup loads and then automatically adds additional channels because it sees those connections. I have to delete them. EDIT: Clarified the first paragraph; added Delay to the DSP that would show up in monitors.

Watching videos or reading examples about the uses of Dynamic EQ don't really make clear exactly how the process works. Neither does the User Guide. So I ran a controlled test with an audio oscillator as a test signal. 1. When the signal level is below the Low Threshold, the Low Gain is applied (either boost or cut). 2. When the signal level is above the High Threshold, the High Gain is applied. 3. When the signal level is between the Low and High Thresholds, neither gain is applied. Simple and concise. Now this makes total sense to me. An example: We have a singer whose voice gets boomy when he sings quietly, but thin when he sings loudly. This is a result of his mike technique: he "eats" the mike when singing softly and backs away when he belts out loudly. Because most cardioid mikes boost bass close-up, his working distance causes the bass to vary. How do we smooth out his tone? 1. Locate the "boomy" frequency by listening (or sweep a standard EQ to temporarily boost till you find the offending range). Let's say he booms broadly between 100 and 250Hz. 2. Set the Frequency to the center of the offending range: 175Hz. 3. Set the Filter Q fairly low, to make the notch wide enough to cover the entire range from about 100 to 250Hz. 4. Set the Low Threshold to a level where the singer is singing quietly and close (he'll have to sing for you to find this level). 5. Set the High Threshold to the level where the singer is loud and backed away. Again, this will take a bit of singing to find it. 6. Now set the gains. Set the Low Gain negative, to remove some boom when he's quiet and working close. Set the High Gain positive, to add some boom when he gets loud and backs away. Adjust both gains to get an even tone at both levels. Attack controls how quickly the Dyn EQ acts on the signal; Release controls the length of time the Dyn EQ holds its Gain actions in place, before "resetting" to check the signal's level again. Now the singer's voice should retain an even tone at both distances. (But if he changes mike types or singing distances, your settings will need adjustment.)

From page 2-16 in the manual: 1. In the Stage View, select the Stage Icon whose preset you wish to replace. 2. Select a preset category using the Category menu located in the top-right of the dialog. 3. Select a preset from the preset list in the right side of the dialog. Decide if you want the Global FX settings for the new preset to be used. If so, enable Include Master FX. 4. Tap Load Selected. That being said, there is a list of all DSP presets by category (available here), but I haven't found a better list that says which preset type each preset uses. (See Appendix C for the available DSP processing chains.) So you'll have to keep trying different presets till you find the DSP chain you want. To do this: 1. Take a good guess at which preset will have the DSP chain you want. I'm assuming most drum and some vocal presets have a gate. 2. Load a preset. 3. Go into Tweak mode. The DSP processing blocks will be listed at the left. If you want to change presets, tap the folder icon at the top to get the Load Preset dialog to try another.

The Global FX D is a stereo delay, but the other effects don't say they are. The only mention of "stereo" in the user guide is when dealing with inputs, and the FX D. There is an output preset for "In ears (stereo)." I don't know if that will provide a different stereo image than "Monitor (stereo);" it may only have different EQ and limiting.

As SiWatts69 said, a split-second power dip (essentially a spike below normal line voltage, not over it) can disrupt a digital system. You may never see the power light flicker, but power-supply voltages can quickly drop. mncanuck, you might run a test by plugging all the speakers into the same outlet (even if you need extension cords, it's OK for a test). I have seen many electrical outlets wired using the push-in contacts rather than using the screws to hold the wires. Those push-in contacts have a very small contact area on the wire and they oxidize over time. An outlet that worked before may begin to develop momentary problems. Also, be advised that most power conditioners are simply surge suppressors, not unlike those in a power strip. They don't regulate the line voltage at all.