zolko60

That is extraoridinary claim especially when people use IEMs which vastly reduces the in situ conditions influence. Well... this is art, not science so extraordinary claims require no evidence or a banana anegdotal evidence is convincing :) I am waiting for some yt video titled "Why leveling the presets is only effective during a gig sound check" ;)

DMX cables work well and they happen to be cheaper.

If everything is so subjective and eviroment dependent as Digital_Igloo claims than "offline" leveling hardly makes any sense. If leveling does not mean setting average percieved loudness apart of enviroment, what corresponds to some arbitrary LUFS value, then our disscusion is about taste and feel. The are two methods of leveling the preset loudness, right? The first is using ears, the second is using sophisticated K-weighted loudness meter. Everybody can try both of them and compare the results. One can even use metering first and correct it by ear. If the claim is "Some preset loudness is 4LUFS louder than another one but the percieved loudness is just the opposite" - well, it needs some evidence. I can not just choose to believe in this. I use my ears for leveling but I observe people ask about metering. K-weighted metering is the only loudness metering that works. It is now used for broadcast and streaming because it makes leveling as precise and objective as it can be.

If the meter is telling me Preset 01A is 4LUFS louder than 01B it is. If I feel 01A is lost in the mix what can I do about it? - Use Preset 01B which is set for different purpose? - Set my monitor louder? - Come closer to my monitor? - Ask monitor guy to set my monitor louder? - Change the venue? - Lower the humidity? ;) Yes, but we are not talking about percieved loudness of presets but leveling them. Leveling in this case means settiing them to uniform level, not to mix context. K-weighting and windowing are the most precise methods so far. If leveling is subjective, venue, monitor, temperature, humidity dependent so relying on ears does not bring anything to the equation or brings that important factor which can't be done away of the particular sitiation anyway. Maybe an electic gutar needs G-weighting (to be invented) and guitars tuned B dropped needs another weighed meter... OK, enough. Use your ears! :D

It is. The unity gain is when each input/output pair is set to instrument level with no block or Volume Knob (attenuator). When you use instrument level for an input and line level for an output you can find 8dB attenuation spot with Volume Knob what corresponds to instrument level with lower S/N ratio but larger headroom. Please consider that eg some clean boosters have 20dB of boost. If your headroom is 10dBFS at Helix input, what is the case of peaks of PAF style pickup, you are not able to achive such amount of boost simulating that booster not distorting the DA converter unless you set higher DA "gear". If you use amp simulations the unity gain is irrelevant - only interfacing with other line level devices. In this case I like to know what is my full scale "gear" is. The ideal situation is when both digital units has the same, eg 0dBFS=18dBu which assures you can controll or monitor the headroom on one device only.

The link lands in the spam folder in google mail. I have no success in getting link (in the video description) for those but working on it. ;)

The Volume Control is manipulator for the digital attenuators before DA converters it is controlling. I was given that info by Customer Service. Fully cockwise if the input/output pair is set to the same level (instr or line). 0dB Because Line6 probably doesn't think this is important enough to be published. You will find partial spec here: Full Scale levels according to some unofficial measurements are 19dBu for line level and 11dBu for instrument level.

Meters do help in leveling presets or anything that needs to be precisely leveled. There are very good free K-weigted loudness meters of Youlean or TB Pro Audio so if you use DAW and don't trust your ears feel free to use them.

You can aggregate Helix and Audient on OSX (high latency, direct monitoring only when monitors are connected to Hx) or use SPDIF (eg amp left, DI right)

Monitored signal should be the same as recorded. The only difference can be the volume. The recorded (monitored) volume depends on your track faders, master fader, Hx trim setting. You can compare them by switching direct monitoring to DAW monitoring while playing (I know - 37ms latency is painful) 10-15dB of headroom is normal in case of "amplified" signal - you can still adjust it in Helix blocks. 20-30dB of headroom is normal in case of DI signal.

You can use "Impedance Matchers" which basically are reversed passive DI boxes (1:9 step up xformer, 20dB of gain), but Shure SM57 "amplified" this way comes after instrument level input and AD at about -30dBFS peaking while hammering its capsule, so 20dB of gain has to be done digital eg. by adding two 12dB gain blocks . Condenser mics would be fine, but require Phantom power, so in practice some mic pre is a better solution.

Helix MIDI Guide comes in handy. (Attached) The Unofficial Helix MIDI Guide.pdf

The impedance thing can be a suspect, but after a Kemper or any buffered stomp - not likely. The only effect dry path simulation I know about in Helix is the Cosmos Echo with a Dry Thru parameter set to off, but since all Hx delays are heavily low passed the impression could be they affect the tone.

Sure it is. Helix can output both processed signal and DI to DAW either with USB or AES/EBU. Reamping with tube amp usually needs some galvanic separation and/or attenuation beetween interface output and amp input. If you have real speaker, no load box is necessary but DI box beetwen amp output and speaker (of course if someone wants to take advantage of IRs).

Maybe you miss the "Mode" switch pressing to change footswitches functionality?

I do not prompt anybody to connect speaker output only to a DI box. A load IS NECCESSARY! The best load is the speaker. Di Box is connected paralell. Amp out is connected to DI In and DI passes signal to speaker by DI Thru, Technically speaking load boxes are artificial loads plus DI boxes. Torpedo can be used also as DI. Lots of amps used to have DI outs built in. They were useless some time ago because analog speaker simulations were poor. Now when we have IR technology we can at last take advantage of such a possibilty of interfacing. There are some drawbacks of course - if you monitor with real guitar cab, all post amp effects must be routed to another monitor. I believe 4CM method is better for live use and DI better for recording. Di boxes are intended for a lot of purposes. I have couple of them. Samson S-direct has even switch label Guitar/Speaker. "Speaker" adds resistor to ground therefore pads and lowers the input impedance. It does not matter much. I found some schematics: http://sound.whsites.net/project35.htm Still if somebody is not sure what he is doing and wants to record completlety silent having nice DAW IR loader full of IRs so reactive load is worth considering. Torpedo 300$, UA-OX 1300$.

Sure, safer but when I connect preamps to IRs guess what... they sound like there is no power amp attached. :D My friend claims that the real tube amp sound is all about real power amp overdrive not preamp overdrive fart. ;) So there is a better method - connect a di box between amp and guitar cab and steal the full fet yummy tube sound from there to Hx FL/RA mic input. In case of LT/ST an isolator to line level return shoud work as well. Your tube power amp is loaded with speaker. Di box is 30$, reactive load is 300$. I recommend passive transformer one for this purpose. Better isolation and no phantom power needed. While isolators/separators are usually 1:1 unpadded, some math is needed: Lets say Hx lline input is 6V at full scale and.your cab is 8ohm. P=U^2/R. Max power to get full modulation can not exceed 4,5W (it is still quite loud). So to get 100W (30V@8ohm) DI connected to line input will work just fine. Again, nothing harmul but distortion at AD converter will happen because its input imedance is too high to draw current. Can your mileage vary? ;) You definitely have to load tube power amp for power tubes and output transformer safety (!) but also for the sound. No load or resistive load instead of reactive - no impedance cuve that shapes the power amp frequency response and harmonic distortion.

IR workshop is a program for IR tweaking. You load impluses not with File/Open but in "Impulse" screen by "Open Sample". Lancaster Audio Pulse is rebranded version of Ignite Amps NadIR. NadIR v.2.0 is a dual 500ms IR loader. Unlike mixIR which has one 8s stereo engine, it can mix two IR paths , make independent LPF, HPF and resonance adjustments, can not save mixed IR. All DSP operations of mixIR are convolution based so if you want to make any EQ you have to load IR to serial IR slot.

Right. Low gain ones are even hard to push to the unity gain. I reported it as a bug.

Nice! I wish you made some IRs. I think it would be awesome to share something to play not only to listen. ;)

In amps models an impedance curve is simulated which induces a huge treble boost. If you go to a tube power amp that boost double. 1. Maybe try preamps instead. The impedance curve is not present in preamps models. (Good sound for you, worse for the audience) 2. Use 4cm method and mic your real cab. 3. Reverse the impedance curve using some eq. 4. Steal your amp output back to Hx using transformer isolator before the speaker. 5. Get rid of your amp and get some wedge or/and IEM instead. Or just use any combination of above keeping in mind the Stomp is limited to 6 blocks and maybe producing similar sound for yourself and for the audience is not possible with the real amp/cab.

1. Yes, correct, 2. No practical workaround.

Yes I understand what you mean. You have spend 200$ for DSP converter and this is probably the best solution. I have been connecting AES sources to SPDIF receivers using simple adapters for 20 years. For up to 3m cable runs I don't care about cable impedance or matching transformers. SPDIF to AES/EBU should be even simplier, but if you can afford the best, safe, stable solution I am happy with you.

In case of "low latency monitoring" and "direct monitoring" no DAW monitoring should be heard Delay compensation state is irrelevant for direct monitoring. It should not be. Simple test is to record something recorded via DA/DA and compare if there is any offset. No. It should be constant for a buffer setting for the same interface. I also hear and feel delay with DAW monitoring at 20ms RTL of Helix as an interface. I do not hear latency with 2ms low latency (Hx direct) monitoring. PT is not recording any latency in both cases. It auto compesates unless there is something wrong. This is the issue of all audio interfaces. There are two ways of monitoring. In case of Helix RTL is quite high so direct monitoring is a must. I use three interfaces with PT. Until i adopted SPDIF recording I used to record guitars with Helix at 256 samples buffer. It is 1651 samples 34ms RTL. With no auto compensation those tracks would be completly off beat. With my other interfaces I could work with the same heavy loaded sessions with 128 samples buffer what gives me about 12ms RTL. I can play with 12ms daw monitoring latency. It is like having guitar cab 4 meters away.

I make IR responses sessions right now using Pro Tools using analog loopback. I can assure you PT usually gives a sample precision auto compensation during recording. My interface has 606 samples RTL at 48kHz 128 sample buffer and one sample signal recorded with an analog loopback sits exactly where it should. There are some exceptions - sometimes a driver doing something strange and there is full RTL delay recorded until a reset. The second issue is bussing the track to another track/aux which deliver additional latency. It is discussed for years as a bug because that latency is not corrected with PT latency compensation. If you can describe at what circumstances your recorded tracks are latent maybe I can replicate what you report, or maybe you can do some simple analog loopback tests to make sure you are not wrong. Setting I/O insert shows full RTL value in latency monitoring indicator under the channel strip, but you are not talking about RTL offset but some offset you claim to be present in recorded track.