Rewolf48

It is an analogue signal from the L2t output, so a microphone cable will do fine, but it is line level.

They do look great and with the conversion neck I decided to have a look myself, but I really can't justify $400 on a neck

The free trial requires you to be connected every time you use it, the licenced version requires that you are connected only the first time to verify the licence and register the computer - after that there is no requirement for it to be connected.

I feel obliged to update my earlier post after a more successful session doing a band recording. We were recording the full drum kit in a separate room using 7 mics (o/h * 2, kick, snare, hats, toms) with the bass (direct + Helix Amp) and a guitar tracking in the control room with everybody playing together. So 10 channels on the M20d had to be be recorded simultaneously into Reaper with me acting as engineer. Previously attempts had been very difficult due to the latency, but this time I persevered and discovered a couple of offset settings that made life a LOT easier. In Reaper Preferences as well as the Audio Device settings where you pick the ASIO StageScape M20d and the input range (I only used 1 to 12) I discovered the Recording settings where you can apply offsets for Output and Input audio driver latency. With these you can align the playback and recording so the Reaper aligns the tracks for you automatically. The process I followed was: 1) Record a click track, then play it while hitting the M key in time to what you can hear - this places markers on the timeline, then you measure a time selection between the click and the matching marker and enter this as the Output manual offset. This will align the displayed audio with what you hear - repeat to verify and tweak if required - I was able to get the click on the audio aligned with my markers using an offset of 830ms. [yes output latency really is that bad] 2) Now plug a microphone into an input channel and record the click track coming from your speakers into a new track. Measure the time delay between the two and enter that as Input manual offset. This will shift recorded tracks forward by this time and means that newly recorded audio is automatically in time with previously recorded audio. Again verify and tweak - I was able to get the recordings aligned to a single waveform using a 41ms offset [not great but not that bad] Having made these changes there was still a huge delay between hitting play or record, or moving a fader or changing eq, and it affecting the audio you hear which means I couldn't use it as a studio interface for mixing, but at least any recorded audio was immediately aligned and we could simply listen back to the takes without any faffing around. Line 6 - please give us a "Studio mode" where the mixer simply acts as a low latency 18/20 In and 6 Out USB Audio Interface...

Stacked on top of each other? There isn't a hole on top of an L3 for mounting another and I doubt that it would be stable if you are relying on the base of the lower speaker. The recommended set-ups for bigger concerts would be to mount each pair side by side on top of L3s Subs as per the diagrams in the Quick Start guide using the poles to link them. https://line6.com/data/6/0a06434d9ac8502007180aca2/application/pdf/StageSource%20L3%20Quick%20Start%20Guide%20-%20English%20(%20Rev%20C%20).pdf See Page 22. There are probably some music styles that don't need the extra low end, but most will benefit from using the dedicated subs to extend the range allowing the L3 to handle the mids. It might be worth considering a pair of L3s to extend your L3m before getting more L3m. I can't see anybody with them in stock either, but that doesn't surprise me that much - I very much doubt that they are a fast moving item. Bax and Andertons say that they can order and deliver quickly, but that depends on Yamaha Europe having them in stock and responding quickly.

Yes people have tried it and some say that effects such as sprint reverb sound much better between amp and cab - the difference is that eq is applied by the cab to result of the effects. The potential problem is that stereo effects (if you are using them) will be collapsed back to mono by a single cab/IR block but should be good with a dual cab block (dual IR doesn't exist - you have to split the path to get a stereo IR)

You can try voting up the suggestions for the ability to blend the piezo pickups with the acoustic model - that way you can go 100% piezo output and potentially input that into acoustic guitar IRs in Helix or other IR loader too. https://line6.ideascale.com/a/idea-v2/515946 https://line6.ideascale.com/a/idea-v2/576212

I tried this yesterday - I downloaded soundcamp onto my Galaxy S5Neo and got an OTC cable. When it was all plugged together the phone could definitely see Helix and Soundcamp / Samsung Professional Audio was grabbing exclusive access to the USB Audio device. But no sound output via USB, instead it used the built in speaker. When I unplugged the cable it actually said USB microphone unplugged, nothing about a speaker, which is consistent with what it was doing. I did a bit of digging around, and while not Professional Audio specific I did discover that the USB audio built into Android 5.1 onwards has the following spec: The problem I believe is that Android USB audio is intended for simple USB speakers - mono or stereo; it can't cope with Helix 8 channel output, and despite the description it doesn't do the decent thing and just restrict itself to USB 1 & 2 - you get nothing at all. I will keep plugging at it - I only want to be to trigger samples using Helix pedals (like Alex Lifeson does) for intros and special effects, but at the moment it has to be my laptop.

Line 6 charge local sales taxes based on the address you enter. So in UK it is $99 + 20% VAT then divide by 1.3 to convert to GBP. In the US it varies as sales taxes are local, so in Chicago it might be $115 after 15% Sales Tax but in Alaska it is $99 because they don't have Sales Tax. US Prices are always before Sales Tax except vending machines where it would be too confusing.

Glens patches include a custom tone matching IR at least in the Boutique patch sets - an IR is a method of embedding potentially very complex EQ curves into a single unit, so Glen has tweaked the base amp to get close to the target amp tone and then created a tone-match IR to get the last little bit. That is the community view anyway, Glen himself didn't say, but you might be able to get him to explain if you ask him directly - he is gangsterusa on these forums

That can work but is more complex on PC - most software can only address a single ASIO driver, but apparently (because I haven't tried it) ASIO4ALL can act as an ASIO Hub, so you connect to the individual ASIO drivers in ASIO4ALL and then connect to ASIO4ALL in your software and gain access to all inputs and outputs. According to one user on the forum he has 28 inputs and 8 outputs working across 3 devices at 96khz with 128bit buffers: http://tippach.business.t-online.de/asio4all/phpbb/viewtopic.php?f=6&t=1109&sid=2e701cea0a18500b6e79f1e69b2c9bd9#p1219 This would allow you to use USB 7 from Helix as direct feed into Native and have Helix patch output in via USB1/2 at the same time and send the mix out to the Presonus. It has got to worth trying this just for the flexibility.

Have you tried the S/PDIF output from Helix into the Studio 192?

It is only online for the demo version to stop you downloading the demo and then taking your computer offline and resetting the date. The purchased version is fine offline once registered on up to 4 computers and unlimited instances running at the same time.

Simplest solution is Minidisk player into M20d Aux Input without doing any conversion and just play it. Or copy to phone/iPad/mp3 player/comuter adn do the same. More complex is to copy tracks as mp3 to the SD card and setup a playlist. Having done this you have limited control over start/stop and next track using the footswitches. It works fine if the tracks are set up in the required set order and are always played in full. Very complex - and only worthwhile if you need the ability to mute or control the level of individual tracks - is to create a set-up for each individual instrument and record a few seconds of nothing then replace the individual tracks with stems in a folder with each track named "N_channel.wav" where N is the input number and channel is the name on the mixer. This will allow you mix and mute the tracks along with real inputs, but offers no real control across songs. It is very useful for one or two person missing rehearsals as their previously recorded parts can be played along with the live parts.

You can use a Send without killing the path, so a single send block is all that is needed. Or if in the digital domain USB 8 is the Microphone input direct out pre-everything. DI answers this question in this post and it is elaborated in some following questions and answers: http://line6.com/support/topic/14264-helix-faq/?p=114985

I also have the Novation Backpack which was about half the current price on Amazon. It does hold a Helix comfortably and has space for a reasonable amount of extra gear but it is really intended for a lighter bit of kit. My use is house to car, car to rehearsal/gig and back again so it is not as if it has to hike over the Rockies and it will do me. But a lot of the decision was based on the "you spent HOW MUCH????" reaction to the Helix purchase and if money wasn't an issue the official one was definitely the preferred option.

Would this UK person be Mark Wright? - I think you mentioned him in another thread: http://www.markwrightmusic.com/home.html

I also use ReaTune. Yes it is a real-time tuner for tuning, but it is also a pretty good and subtle pitch correction for vocals, and is even used internally in Reaper when you slowdown or speed up tracks to maintain correct pitch (ideal for practising complex parts) http://wiki.cockos.com/wiki/index.php/ReaTune Unfortunately the logic uses the core engine and it isn't available as a separate download (like some of the others as the ReaPlugs package). There are plenty of others if you DAW doesn't include one - e.g. http://www.gvst.co.uk/gtune.htm Using a separate VST/AU/AAX means that you can have it active all the time

Are you saying you missed the mega-thread on The Gear Page? Because it was all over there too. Line 6 have been completely open about this from the moment Native and LT were announced respectively. And you did some research before you spent the money on LT - right? This forum is not exactly hiding - try any search in google about anything technical about Helix and this forum comes out on top. I personally might have complained that I bought a Helix Floor 2 months before LT was announced... for less money, or about how Yamaha Europe then dropped the prices a few months later, but actually I have nothing to complain about. I bought something that I thought was good value for me at the time (and spent more than a year justifying) and any subsequent pricing changes is just what happens. It doesn't invalidate my original decision. £120 is still more than double what I have paid for any single item of music software (UK VAT at 20% on top) but once I get back into the UK from this work trip I am going to make sure it works on my PC with the free trial and probably cough up the extra cash

Just hope it isn't on a Windows 10 machine and Microsoft force an update with mandatory reboot when you are not looking.... :angry:

I did send a message about this thread to Frank Ritchotte (Operations Manager with responsibilities for Support) as there appeared to be a huge mismatch in how things worked in the US and how they worked in Europe (as a European it was in my interest). Not claiming any credit here but it is possible that there have been some changes made as a consequence.

You have the choice as to whether you connect the M20d directly to the MacBook or not - I suggest not and that you get an additional audio interface. The big problem with the M20d is that while it does support audio over USB for all the channels it unfortunately has dreadful latency (1 second+). It is possible to connect over USB and directly record all of the inputs directly into Logic, but it is just not possible to use it for overdubs (additional tracks) without being driven crazy: you set up a channel to record and play back what you have previously recorded and record a new track and the misalignment is huge - you have to drag the track back and try to align it just to hear what you did; and that absolutely kills the session. I know because I have tried it. I suggest you connect all of the gear into the mixer and the mixer into some speakers (or use headphones) but you should feed the instruments you want to record into another audio interface (use the monitor outs for example) and the output of the audio interface into a couple to channels on the mixer. For recording a band all playing together you record onto the SD card and then read the card directly using the laptop.

A guitar pickup is an electromagnetic noise detector, put that through a lower quality microphone input on a computer which adds electric noise, at 16 bit resolution and you are starting with a noisy signal. Then feed that signal into a high gain amp model (which amplifies it all!) and you will get a lot of hiss and noise. [forgive me if not all of these are true, but you haven't given much information] But you would have noise if you had the real amp present as well - it has less to do with the model and everything to do with the quality of the signal chain, the fact that it is less audible when there is no amp sim simply means that you are not amplifying the noise as much.To remove noise you need to address all aspects: Humbucker pickups cancel a lot of magnetic noise compared to single coils, but even with them try to move away from electronic noise sources such as computer monitors, WiFi/Bluetooth transmitters, dimmer switches, strip or cheap LED lights. Is the guitar electronically sound? Switches, sockets, cables? You don't say what you are feeding the guitar lead into - but there is a huge range of audio interfaces and the lower-end especially built in stuff will be adding more noise. HD500 or Helix as an interface is much better as are the more expensive studio quality audio interfaces. They will capture the input at 24 bit resolution @ 44KHz (or higher), but even here you need to watch gain staging (how loud the signal is at each stage - too low and the later make-up gain adds noise, too high and you get clipping). Gain Staging also applies internally in cubase. Amps generally need much less gain than many think, but if you really want a high gain modern metal sound the most important component is the noise gate normally placed early in the chain. And then there is gain staging the output side from Cubase to your ears - that also requires proper gain staging; turn you power amp to max and then feed a very low signal in an the noise will come from the power amp. To see how much is coming from the Amp model and how much from everything else simply try a very clean amp model or to simulate a high gain amp mute Guitar Rig and add an EQ block (flat) and turn the input and output levels right up.

MIcrosoft won't let me go back to Windows 95 either... perhaps for a similar reason ;) It is really old and contains various bugs that they have since fixed, so they only let you go to the versions they are comfortable supporting. The only way to go to 1.71 is if you can find a copy of the firmware file that somebody has saved on disk from back when it was current - I am sorry but I don't have anything that old. All guitars are unique: different tree, different age strings, different components (within a specific tolerance), they will sound different - even variax guitars. You can probably tune the models to be close using Workbench HD (start with global string levels as they have a significant effect).

It is a good idea as long as you work out the practicalities. The biggest problem for you with an IEM such as the SE315 is that it blocks out all of the other noise and you are very isolated; due to psycho-acoustic effects using only one is not a good idea, so you really need a way of getting some monitor mix and/or stage sound in as well. Unless you really needed a 20db reduction in level because the stage volume is way too loud. With the Helix there is no reason why you cannot mix another input by dedicating say path 2B to a mic input or monitor feed. The other problem is the cable - having two flying leads means you will get tangled or you can tie the headphone cable to your guitar lead which makes that stiffer and is a problem if one or other lead needs replacing mid gig. For everybody else they now have to monitor you through another means, which may be a general monitor mix; unless everyone is using IEM in which case you would be using that system.