zolko60

14 hours ago, Digital_Igloo said:

Correct. You have to level your presets in situ.

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.

3 hours ago, cruisinon2 said:

So unless you really enjoy constantly having to adjust volume in every different scenario you find yourself in, setting every patch to some arbitrary dB value accomplishes what exactly?

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. 

3 hours ago, Digital_Igloo said:

Say you're in the studio, and the meters tell you Preset 01A is 4dB louder than Preset 01B. You get to the gig, and now Preset 01B sounds way louder! What gives?

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?

;)

4 hours ago, Digital_Igloo said:

Dozens of things affect perceived loudness of presets (even temperature and humidity!), and meters account for only one of them.

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

1 hour ago, tahiche said:

With no blocks input from guitar and output should be at the same level. From there I can work properly. If it's not right I have to go through every path adjusting everything again. 

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.

On 5/23/2019 at 4:03 PM, zolko60 said:

I had no luck to buy that free pack from their store (no link sent) but maybe it will work for somebody.

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. ;)

On 11/24/2018 at 3:38 PM, zolko60 said:

So what is Volume Control?
Digital attenuator/booster before DA converter which I can assign to Multi Output as a master fader for XLR/1/4/AES/USB1/2out or as master fader for individual outputs (after their faders in Output Blocks)? Where is its unity gain setting? What is it's maximal boost? Why have I to find it out by experimenting? I know how few people read manuals, but I would appeciate adding couple of sentences more.
BTW. When can I find specs like Input/Output impedancies, Full-Scale Input Level for Helix LT?

The Volume Control is manipulator for the digital attenuators before DA converters it is controlling.
I was given that info by Customer Service.

On 11/24/2018 at 3:38 PM, zolko60 said:

Where is its unity gain setting?

Fully cockwise if the input/output pair is set to the same level (instr or line).

On 11/24/2018 at 3:38 PM, zolko60 said:

What is it's maximal boost?

0dB

On 11/24/2018 at 3:38 PM, zolko60 said:

Why have I to find it out by experimenting?

Because Line6 probably doesn't think this is important enough to be published.

On 11/24/2018 at 3:38 PM, zolko60 said:

When can I find specs like Input/Output impedancies, Full-Scale Input Level for Helix LT?

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?

2 hours ago, rd2rk said:

I'm a pretty simple guy, and have a pretty simple rule on this subject. If the jack says "SPEAKER OUT", it either gets connected to a speaker, or to a reactive load box or other device that's INTENDED for the purpose. I've never heard of a DI that's DESIGNED to take a SPEAKER OUT. Though they MAY exist, I think they're called LOAD BOXES or ATTENUATORS when they're INTENDED for that purpose.

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$, UA-OX 1300$.

4 hours ago, kirias said:

My head amp has a pre amp out as well. What if I connect the amp through this out into Helix return and use IR? Would this be safer?

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? ;)

14 hours ago, kirias said:

Hi! 

I've connected head's speaker out in Helix Return so that IR works as a cab and it works fine. Actually sounds awesome and even the effects sound great as well. Is this harmful for the unit? Damn sounds so good to be dangerous.. What do you guys think?

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.

1 hour ago, zenguitardude said:

So, if I am recording and I leave the track armed and unmuted, I can hear a difference between the direct signal and the channel signal. It can vary from a Phase sound to an actual slapback, depending on the latency/delay.

In case of "low latency monitoring" and "direct monitoring" no DAW monitoring should be heard

1 hour ago, zenguitardude said:

I have delay compensation and low latency monitoring turned on.

Delay compensation state is irrelevant for direct monitoring.

1 hour ago, zenguitardude said:

On playback that performance is delayed. It is obvious sonically and visually.

It should not be. Simple test is to record something recorded via DA/DA and compare if there is any offset.

1 hour ago, zenguitardude said:

The delay amount seems to vary. This is the nature of latency.

No. It should be constant for a buffer setting for the same interface.

1 hour ago, zenguitardude said:

I hear and feel a delay between when my hands execute the performance and the PT rig is recording it.

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.

1 hour ago, zenguitardude said:

I think it is a Helix issue because the recommendation on how to record is to monitor THROUGH the Helix as you record, because that monitor path for audio is PRE USB send to the DAW,  so that you don't experience the latency as you play.

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.

23 hours ago, zenguitardude said:

The topic I cannot find is:
after addressing monitor based latency, in other words hearing latency as you play/record, no one discusses how to fix the actual recorded latent track.

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.

22 hours ago, zenguitardude said:

not a way to measure and offset, that i know of.

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.