Display MoreServus,
hab gerade versuche via SPdif zu recorden und es treibt mich in Wahnsinn.
Ausgang Kemper -> SPDIF out -> Focusrite 6i6 SPDif Eingang
Kemper SPDIF Lautstärke voll aufgedreht und SPDIF Output auf Master Stereo. Es kommt einfach kein Signal beim Focusrite an. Jemand eine Idee?
Ist das Audio Interface auf "slave" bzw. external clock eingestellt? Ansonsten geht es nicht.
Most likely at least any of the other bands will be using a Kemper. Since you are a nice guy i am sure you will be allowed to use it...
Bin auch älter, aber irgendwie kann ich mir diese silent stage am Besten noch mit E-Drums für eine Rockband nicht wirklich vorstellen. Passt eher zu Tanzmukke, wo ist da noch RnR frage ich mich.
Fragen kannst du dich das, aber die Antwort erhälst du erst, wenn du es ausprobierst. Soviel "Rock N Roll" wie mit dem Silent Stage Ansatz hatte ich in den letzten 25 Jahren nicht mehr. Jeder kann alles perfekt hören, und vor allem bekommt das Publikum einen viel druckvolleren und wirklich differenzierten Sound. Die Vocal Performance profitiert unglaublich durch die ganzen Möglichkeiten bzgl. Kompression usw.
Ausserdem fallen eben genau die vielfach beschriebenen Probleme weg, dass Kemper Rigs live plötzlich nicht mehr funktionieren. Der Grund hierfür ist nicht die höhere Lautstärke, denn auch eine gut produzierte CD klingt live bei 120 db auf einer gut eingestellten PA immer noch genauso gut. Das Problem sind die anderen Instrumente und die Übersprechungen, besonders bei den Drums. Sobald ich diese Probleme eliminiere und alle Instrumente so weit wie nur irgendwie möglich live genauso klingen wie bei einer Studio Produktion funktionieren logischerweise auch die Gitarrensounds genauso gut wie zu Hause in einem Mix über Nahfeldmonitore.
is the general consensus that triggering a plugin like SD3 or SSD from e-drums the best way to go?
I personally hate the Roland drum sounds and would rather spend more money to get the best possible sound!
It will provide the best possible sound quality IF that library does provide the sound that you are looking for. However you need a quality Notebook + Interface and have more possibilities of failure in a live situation.
The alternative would be to get rid of the TD-25 module and replace it with a used 2BOX module. You should have some coin left this way for your monitoring solution. I have all the sounds from all Superior and EZDrummer Expansions, BDF, Slate and other samplers converted and ready to use, probably like 250 high quality snare drums, most of them with over 40 velocity layers, while most Roland modules only have 3 or 4 velocity layers.
The linked article is measuring not only MIDI latency but the overall latency from hit to sound ... so it includes the entire processing and DA latency on top of the pure MIDI latency.
Also they used a buffer size of 128 samples. On top of that it's 7 years old. We've improved a bit since then, hehe.
The typical MIDI over USB latency should be somewhere in between 1 - 2ms (at least for the module in question here, TD25)
Add the 3ms of the entire processing and DA and we end up with ca. 5ms .... and let's grant a tiny bit of latency for the monitor DSP, we'll likely end up at 6ms total which is not an issue for any drummer.
In a 120bpm song, a latency of 6ms corresponds to less than a 1/1024th note
No, it has nothing to do with audio
only the difference between the Mic PAD and the recorded MIDI event is of importance. I measured those myself, and believe me, no drum drum brain in the world has less than 4ms. But probably i could talk this forever. All i say is that adding 2ms to 4ms od module latency is of no concern, but adding those to 10ms or more is critical.
No friggin' way, any MIDI drumkits have a MIDI latency this high. Maybe you're talking audio but that's not necessary in the OP's use case.
No, i am talking about pure drum hit to MIDI conversion. I lost the link with a comparion of large number of drum modules, but check this out (sorry, only in german):
https://www.amazona.de/test-alesis-dm10-pro-kit/7/
Whether a drum module is newer or older does not not necessarily mean that they do the MIDI comversion faster or slower, it just depends on the model. Old DDrum3 and D4 modules are even faster than the TD-30, though Roland modules mostly are pretty fast, some of them below 5ms. The 2box are 7ms, Some Alesis are over 12. Some cheap Thomann stuff even was nearly 20 ms.
And the latency can't be felt, promised. The "MIDI IN ->SD3-> Audio Out" setup doesn't have the same roundtrip latency you experience when going "Audio In->DAW->Audio Out".
This is too general. I habe been doing this for like 15 years or so. The latency of the audio interface always comes on top of the latency from drum hit to MIDI out produced by the module itself. For example ,the Alesis Crimson modules have a native latency of about 12ms, which already is too much. Add another 2ms from the interface and this is unplayable for timing critical stuff. Some Roland modules like the TD-12 are about 6ms, which is a lot better. Each drum brain is different. In our band we don not want any PC or laptops, so we use a 2BOX drum brain which runs all the converted multisamples from Toontrack, BFD, Steven Slate, AD etc. It will not provide all the multichannel stuff and bleeding like SD3 does but it is a very compact and robust system. Plus, the module is very affordable, you just need to add a 32 GB SD Card. We run Kicks, Snares, Toms and Cymbals in strereo pairs into the X32 console for you own IEM mix and then all four pairs are routed out of the X32 to the FOH. Works like a charm.
This is exactly what I meant above ... don't fall for this trap of "lower and lower and even lower" ... it ain't working this way.
The audio processing on the computer dictates the buffer size required. It's very unlikely that you can run a massive SD3 drumkit (and mix) stable using these super low buffer settings.
Yes you can, in fact i was able to use 32 buffers even on older systems stable with SD2 and metal machinery SDX using RME interfaces. Those kits already took up to 2600 MB of memory. I just installed SD3 and have not tested with Edrums yet, but the memory usage is not that much higher than in SD2. Not any preset was over 3800 MB. So i am confident SD3 will work as well.
First of all, switching to edrums is probably the best choice ever to improve the band´s live sound and also the overall performance. Combined with a digital mixer like the x32, Inear Monitoring and the Kemper playing in band feels like a complete new experience. I could not be happier with this setup.
Like stated above, make sure the laptop uses SSDs for the system as well as the SD3 libraries. Also make sure that battery of the laptop is still good in case the power suplly fails.
Using edrums you should consider the lowest possible latencies. RME interfaces will allow for buffer sizes as low as 32. Most RME USB interfaces are pretty expensive. So take a look at the new digiface USB:
https://www.thomann.de/de/rme_digiface_usb.htm
You can combine it with an ADAT A/D converter of your choice. This will make it also a super flexible recording setup with up to 32 Inputs.
Kann am Treiber liegen, oder der automatische Latenzausgleich klappt nicht. Probiere vielleicht mal eine andere DAW wie Reaper, ob es damit klappt.
The ultimate in portability ist switching to in ear monitoring. If i would choose a minimal setup without wireless devives and no backup guitar i could carry all my equipment with one hand. IEM requires having your own peronal mix so in most cases bands bring their own digital mixer. But once you get this done playing like this is a huge step forward.
I did this back in 2012. It actually worked but the results were rather poor.
Display MoreHi,
I am a bit late to the direct profile game. hence my question. I have an old StageLine DIB-100 DI Box. Would that be suitable for making direct profiles? Any tips would be greatly appreciated.
Thanks!
G
Seems suitable since it has switchable attenuation. Use -40 when profiling really loud. Otherwise -20 should do. At least that is what i do when using the Behringer GI-100...
I contacted Mackie since they make the boards that we use in both bands. This was their response (and it makes sense):
"The only way to prevent this is to use the Mimiq on an aux send. The way you have it set up now, the signal is already wet with the effect before it comes into the mixer so there’s no way to remove it with the mixer. The aux’s are mono so you’re always going to get a summed mono. I would use AUX SENDS 1 into your Mimiq, then from your Mimiq back into STEREO RETURNS 1. Use the AUX 1 knob on an individual channel to apply the effect to it. Then use AUX 2 as you monitor send. Aux 2 is pre fader so you will not hear the effect through the monitors."
Unfortunately, that option wouldn't work for me because our boards only have 4 aux sends on them. And the 4 members of the band all use their own aux for their individual in-ear mix. And from what I gather, the suggestion from Mackie would require 2 of those 4 auxes. So it sounds like me buying an extra monitor mixer is the only way to get around this situation. The downside is that while it fixes the problem for the mains and my in-ears, the other 3 guys will still hear the out-of-phase type of sound. If there was a way to use my separate monitor mixer (Allen & Heath ZED-10FX) to get around this and fix it for everyone, I'd be interested in that. But the long story short is, I need 5 auxes and we only have 4.
This is why the x32 mixers are great. There is no fixed number of aux aux sends. You can route as many individual mixes as there are mix buses available. Those mix buses then can be assigned freely to either Aux Outs, XLR outs or even outs on a remote digital multicore breakout box.
Did you update the mimiq to firmware 1.1?
You'd think that would be it. But that doesn't work. I tried just about every combination I could think of. Any time I used an aux send to send ANY signal back to my in-ears, it had the out-of-phase type sound. So I just decided to buy a small personal monitor mixer. It doesn't solve the issue for the rest of the band, but they don't mind at all. I just now run the Kemper as I did before, stereo XLRs to the board. And I take my aux from the board, send it to my little mixer so I can hear the rest of the band, turn the guitar down completely in that mix, take the headphone output of my Kemper using a TRS split cable that goes from the headphones into 2 channels on my mixer, pan those hard left and right, and then run the outputs of that little mixer to my left and right returns on my IEMs.
Maybe your band should consider switching to a used Behringer X32. The smaller versions like the producer sell for like 1000€. Considering the features of this mixer, the price is a joke.
You simply have to use 2 mono aux buses send to pre fader and route the audio channels accordingly. Put the left Kemper signal on aux1 and the Kemper right signal on aux2. Then simply connect both aux out to the inputs of your In ear transmitter. That would be all. If you are using a digital mixer like the x32 the buses can be paired to stereo buses.
One more question. When I've put this in my "x" slot and I'm using the stereo loop. When I'm in the parameters for the stereo loop, I have an option to set it for "pre" or "post". Which is the best place for this? I didn't really hear any difference when I toggled them. I think it MIGHT have sounded a LITTLE better when set to pre, but that could have just been in my head.
This is only important if you really use the mix function of the fx loop. Can be useful for spillover effects. Since you are sending the the signal completely through that FX loop path, it is of no concern here.
Okay, so, I bought the Mimiq pedal tonight. Hooked it up to the Kemper using the stereo loop and tossed on my headphones. Let's just say that I wish I had put on an adult diaper because I'm pretty sure I spooge-stained my favorite pair of jeans. This pedal along with the Kemper's internal effects is an UNREAL combination. I've got a gig this weekend and will test it out in full stereo at gig volume using the PA. If it sounds as good as it did tonight, I'm divorcing my wife, moving to Utah, buying a 2nd Mimiq pedal and marrying them both!
I've never been able to get that old school Zakk Wylde sound from songs like Mr. Tinkertrain, but with this doubler in stereo, with a little micro pitch for a chorus effect, it was as close as I've ever heard without actually layering multiple tracks in a DAW. The idea of having that sound live in a one-guitar band and oh . . . oh. . OH. . . dammit. . . . there goes another pair of jeans!
Thanks to all who recommended this thing!!
Great, have fun 
If Christoph and his team could replicate this pedal as an internal effect i would be over the moon. It seems to process the audio very differently compared to other ADT devices. When you slightly touch your guitar you can hear each touch shift the doubled sound. I got the impression it has some sort of transient recognition going on.
By the way the mimiq also prodocues very interesting results when used on vocal tracks. Since i have my KPA hooked up via SPDIF i recently tried routing a dry vocal track through the KPA with everything disabled but the mimiq and recorded the stereo output.
Thanks guys. I finally got a chance to look at the manual and I see now that the alternative input can also act as another return for stereo loops. So that answered my question.
I'm in a one-guitarist band, so I'm going to give the Mimiq a shot and see how it sounds. Could be a great effect for a small amount of money. And considering I'm not using anything in my effects loop right now, nothing to to lose.
That´s right use both outputs of the mimiq into both the return and alternative input. Set the mimiq to "1 dub", with tightness between 11:00 and 13:00. The levels for dry and effect need to be fully turned up, but the levels were not exactly equal in my setup. So i need to reduce one of the levels a little bit. Also make sure you update to firmware 1.1. This requires a special procedure as listed in the manual. Simply connecting it to your PC will not work.
