jeremyn

The primary use of a lower impedance is to emulate the tone coming out of some old school fuzz and wah pedals. Those pedals have a relatively low input impedance that cause the magnetic guitar pickups to smooth out their frequency response and roll off some of the highs. This can help quite a bit with getting a nice fuzz tone, or a particular distortion tone that relies on the wah/fuzz at the top of the chain to load down the guitar. However, IMO, that also makes the pickups sound a bit lifeless when used in other situations with other effects, especially clean tones. This is why there are true bypass mods for the old pedals, and many newer pedals have true bypass (or a high impedance buffer) as standard. People also put the fuzz/wah pedal into a hardware loop so it gets fully bypassed when not in use. This gives the best of both worlds by allowing the effect to load down the guitar (smoothing it out) while in use, and not affecting the guitar tone when bypassed. It'll be interesting to see what Helix has coming in this area to make the impedance setting a bit more versatile. It'd be particularly useful if some type of 'scene' mode could include an impedance change/block as part of the scene selection within a patch. Then auto could just use what the patch/scene had defined, or be expanding as "Auto, Scene, 1M, etc"

You can't correctly state that the proverbial axe cut into a block of wood is accurate to 1 micron; and for the same reason Line 6 can't correctly state that the Helix tuner is accurate to 0.1 cent. There might be some subset of the measurement system that has excessive accuracy beyond the final measurement result. But, that can only be stated specifically as the given subset. The "tuner accuracy" is the total system measurement accuracy of all relevant subsystems, not the accuracy of a chosen undefined highly accurate subset. Digital tuning is inherently as accurate as the total combination of: the AtoD crystal oscillator stability, the sampling bandwidth/duration, signal to noise ratio, the frequency detection algorithm, detection speed/accuracy/stability tradeoff, post detection filtering/hysteresis, the output granularity. Making the Helix tuner more accurate seems to be as simple as adjusting the display granularity, as that sounds like it's the weakest link in the chain. Although, I do wonder why they chose such a low resolution in the first place. But, I'd be even happier if they added a linear strobe display directly below/above (or instead of) the huge tuning bars. Unfortunately, that is a much bigger change that would most likely impact various points in the software stack. To do properly, the raw signal needs to be re-sampled (digital circular sample rate conversion) against the expected reference note frequency (as detected by the existing tuning algorithm), and filtered for graphical visualisation. None of this garbage where the strobe is just an output visualisation of the tuner's detected note; the strobe MUST be based on the resampled raw filtered input, otherwise don't bother.

Choosing the pan law is an interesting conundrum. It heavily depends on the number and level relationship between the signals that are being combined. 3dB is an industry standard middle ground that is not perfect for every combination of mixed sources, but is better than 6dB or 0dB that might work well for equal level signals that are perfectly in phase, or two signals with wildly different levels. The general rule for panning is that the signals are uncorrelated, and that they are of approximately equivalent power levels.

Accuracy is directly related to granularity/precision. It is incorrect to state that something is accurate beyond its measurement resolution and error tolerance (which includes all intermediary truncations). It is like the old joke, "measure with a micrometer, mark with a chalk, cut with an axe". When someone asks how accurate the cut is, you must include the error margin of all combined actions, you can't just say it's accurate to 1 micron. When I forget to use the custom BFTS mode on my Turbo Tuner I definitely notice it. I can live with the error, but my ear thanks me when I tune with those offsets. Those offsets contribute to the accuracy of the tune, while the precision of the Tuner remains the same in both Chromatic and BFTS modes. This is because the precision of the TT is sufficient to resolve the additional accuracy of using the BFTS offsets. When a tuner has excessive granularity, it is difficult if not impossible to tune accurately (unless you're lucky enough to have a very forgiving ear). Hysteresis between the bars further decreases the accuracy. That means talking about accuracies better than what is resolvable on the display (including intermediate filtering such as hysteresis) has no meaning in the context of using a given tuner. Technically, the precision of the Helix tuner may be very good at the edge of the detection thresholds (ie. the change between one bar and the next may be at some repeatable point with an error tolerance of +/- 0.1 cent). But, if those edges don't align with the centre frequency of the note, then the tuning point must be interpolated between adjacent measurements (which is very difficult considering a non-linear frequency to angle relationship at the tuning peg); therefore, the Helix tuner has an accuracy equal to or worse than the granularity of the display.

Excellent option. Hard to believe they didn't already do that.

If you're maxing out a K12 as your foldback, I hope you're using significant hearing protection. That thing is deafening at maximum volume, which means your drummer is also dangerously loud. If you're competing with a live hard hitting drummer, then you need to get closer to your speaker so you don't have to turn it up more. If you're standing behind your speaker with the drummer in your ear, then you need to move.

This is ideal. Balanced connection, line level to line level, and no chance of phantom being accidentally applied.

The terminology can get confusing. It's best to refer to these things like this: Sample rate: 44.1ksps, 48ksps, 96ksps, etc (kilo samples per second) (can be referred to in kHz, and this is why some people are confused) Nyquist Frequency: 22kHz, 24kHz - (also known as the maximum frequency response, maximum bandwidth) Bandwidth (Frequency response): 20Hz to 20kHz, 20Hz to 40kHz, etc (depends on the analog filters and digital decimation filters in the AtoD/DtoA converters - bandwidth can never be greater than the Nyquist frequency) So basically, the sample rate is slightly greater than double the upper cutoff frequency of the converters. So at 48kHz, you can expect somewhere around a 22kHz bandwidth. Dynamic range (bit depth): 16 bit, 24 bit, etc - (may be referred to in dB, e.g: 96dB and 120dB dynamic range respectively) Bit rate: (More accurately refers to the output rate of a codec when compressing the audio, eg. 128kbit/s mp3. This has nothing to do with frequency response or dynamic range, and may indeed a low bit rate may be able to perfectly reproduce a high bandwidth waveform if the waveform isn't too complex. For an uncompressed raw stream, this would be the bit depth multiplied by the sample rate.) It is generally accepted that the human ear can't discern anything above around 20kHz. (Yes, some small children and 'golden ears' can maybe hear a few kilohertz higher, but most adults are lucky to even get to 20kHz in the first place, and would find those frequencies completely masked by any other co-channel sound.) This has been shown to be consistently true in many scientific studies. The concept of ultra high bandwidths is more about marketing, however, it is does improve some systems that haven't properly de-aliased their internal digital models or simulated distortions. That's why some units sound less 'fizzy' at higher sample rates. Funnily enough, a compressed stream (using high quality AAC/mp3/FLAC/etc) doesn't have a significantly different bit rate for sources recorded at the higher rates. So there is very little (if any) useful or audible information in those upper frequencies that isn't just thermal or background noise, and almost never anything of musical importance. Technical note for anyone interested: Most converters these days are sigma-delta which use a multi-order digital decimation filter and an extremely high sample rate to reduce a heavily oversampled low bit converter into much higher bit depth samples at a much lower sample rate (the nyquist rate). eg. A typical converter uses 128 times oversampled 6.14Msps 4-bit flash converter to produce a 24 bit output at 48ksps, with a Nyqvist frequency of 24kHz (digitally filtered to start rolling off at around 22kHz). The digital filter puts all the conversion noise up into the top of the frequency spectrum which can then easily be filtered away with a simple low order analog anti-aliasing filter. (This is where terms like '1-bit' converter and concepts like DSD (direct stream digital) come from.) It's been quite a few years since improvements to AtoD and DtoA converters have actually mattered for audio frequency reproduction. (Besides areas such as improvement in power consumption, noise floor/dynamic range, and onboard features such as mixers/attenuators/etc.)

Like anything, if you're listening for it - especially if you're the player - then you do notice it. A third party listener that is hearing the entire mix, not so much. The digital cabs don't sound bad, but you can hear a difference when you're A/Bing two mixes, one done with a digital cab and the other a real recording. It's not huge, but then neither is the difference between modelled amps and a real amp. But, since the Helix is all about striving to reproduce a real setup, there's no reason why Line 6 shouldn't at least be trying for the next level of speaker cab emulation.

Only if you're having a problem with humm/buzz when using the unbalanced connections. If you do use the XLR, make sure phantom power is off before you plug in, and the trim is set to zero so you don't clip the input. The XLR inputs on the Bose have decent low and high frequency flat response, so you won't have any dramatic EQ or level problems. Note: always be careful when plugging full range gear into 'mic inputs' (and vice versa) as microphones have different requirements to line level gear. XLR doesn't define the electrical spec for the interface, so there are all sorts of conflicting requirements that different gear have depending on what they expect the interconnect use case to be. XLR inputs don't necessarily guarantee a better signal transfer than a 1/4" and may cause other problems (e.g. phantom power, impedance mismatched, clipped preamps, etc). It's better to match against expected impedance/level (ie. line to line, mic to mic, instrument to instrument in) than the connector type. Balanced is _usually_ better than unbalanced as it rejects coupled hum/buzz and other electrical noise, but you still need to watch for level and impedance mismatches as you don't want to clip your inputs or amplify the hiss when turning up the gain.

Nice board! I bet you've voted on all the Ideascale entries for 'better synths', 'freeze' and 'EHX HOG/POG/*9'.

If I have to, I just use an XLR to 1/4" TRS adapter cable and send to the TSR line input on the mixer. In environments that only have 'mic' inputs, I always use a DI - even if it's just a transformer isolator (with optional pad) from the XLR output on my equipment. That also applies if I'm using a sub-mixer to send to a dodgy FOH where the mixer is locked in a cabinet and only has a bunch of mic sockets available to the musicians. I don't really have an issue with the Helix XLR outputs being 'line' outputs and not DI outs. Meaning, I wouldn't ever expect to directly plug the line output of mains powered gear into the 'mic' inputs on a mixer. I've always applied this rule to my equipment - ie. never plug in anything to an XLR mic input, or any XLR for that matter that is not a known dedicated 'line' input. This protects both the equipment and the FOH PA (desk, amps and speakers). BTW, most small (and large) mixers have both an XLR mic input (where phantom power may not be individually switchable) and a 1/4" TRS line input on each 'mic' channel. On those mixers, only the XLRs provide phantom power. So my rack/Helix/wireless/etc use XLR to 1/4" adapter cables to go to the dedicated TRS line inputs on the submixer. This is electrically much safer with less chance of making a mistake when your tired. Also, no loud bangs/pops from plugging into a phantom powered line while the system is on and not muted. Generally all around safer for any gear involved, including the speakers and your ears. The most annoying situation is when you don't have access to the FOH 'desk' and all you get is a bunch of 'mic' inputs on a wall plate. Think small churches or community halls. Doubly annoying is when these inputs have already been EQed for a typical vocal mic and have phantom power permanently active on the sockets. That usually also means you'll have to 'undo' the vocal EQ with an inverted curve, and hope that the FOH isn't rolled off so heavily that you can't get anything below 150Hz and/or above 8kHz. A small sub-mixer is your friend in these situations - as is the global EQ on the Helix. So, even if the Helix had a super isolated output block, I'd still be using the above procedure - mostly to avoid mistakes with other line output gear. I use enough rack and pro audio gear that have pure line outs that I know are not isolated. I don't want to risk the equipment outputs or the mixer inputs because I was asleep at the wheel. I can understand some people wanting to have an isolated DI output built into the Helix, but it's really only necessary when you are plugging into an unknown input (or a mic input known to have phantom power enabled). And then some would still be complaining about why Line6 didn't use the 'right' transformer (ie. 1:1 vs step down, etc) or a decent quality output transformer (eg. Jensen/Lundahl with a full range and flat response curve), or bigger/better quality coupling capacitors. I'm all for Line 6 having a clean DC coupled line output on the Helix and then using my own high quality outboard isolator(s) if I need to. Although I have nothing against a Helix v2 having an additional transformer isolated DI output, or more ruggedised line outputs, I definitely don't want to give up the high quality full range flat response of the existing XLR line outs. Clean, full range, balanced interconnects are important in the studio.

I've been using these digital systems since the X2 Wireless days. Great products. I used a LiPo 9V in the X2, and I also use a G50 with Eneloop AA rechargeables. They work ok on a small stage without the external antenna, but with the antennae attached they go huge distances. The G10 looks interesting. Could be good for small coffee shop gigs and playing at home.

Love the track, but you're right, the the outro on the HD500X really brings the synthy/verb magic that the Helix version misses out on. The guitars on the Helix version are definitely improved. Either way, both tracks sound great. It'll be good to see some decent synth/freeze & reverb effects turn up in the Helix.

Total speaker breakup at stupidly high volumes where the magnet crashes out is way outside any tonal sweet spot anyway. However, the model should include the nonlinearities of the speaker before it gets to the extreme points. This is fairly loud in the real world, but not stupidly so for most typical guitar speakers. The other thing is the doppler shift distortion of higher frequencies that is dependent on the movement of the cone caused by lower frequencies. As the volume goes up, all these distortions increase. Having a complete series of IRs taken at various dynamic levels doesn't help if Helix can't dynamically choose the appropriate sequence on a sample by sample basis. This is how the Kemper profiler works. It takes a series of IRs that it uses to interpolate between when generating the output waveform based on the level of the input samples. No reason why Helix can't do this too, but Line 6 has chosen the modelling approach where it tries to determine and algorithmically simulate all the physical parameters that have a significant effect on the tone.

Can't you just turn down the input gain on your mixer/PA? Also, it's best to use an XLR to 1/4" balanced cable and go into the 1/4" line input on the mixer if the XLR inputs are all into mic preamps.

The longer the IR, the more frequency resolution it contains. 1024 points is around 20ms/50Hz, and 2048 points is around 40ms/25Hz. So the accuracy of the IR at lower frequencies is much better with the longer IR, and therefore the filter parameters are inherently better controlled. The method of truncating the IR can affect the way it rolls off, and therefore can also have an effect on the low frequency response of the filter.

It can if the output stage of the device isn't AC coupled, or the coupling capacitors aren't designed with phantom power in mind. Not sure if this applies to the Helix, but the fact that turning on phantom power makes the Helix sound noisy is pointing in that direction.

Just in case you missed it, scroll back up to PeterHamm's post (4 posts above this one). As he said, this is dead easy.

No, a single linear IR can't model compression or distortion. But, those extra 'reverb reflections' can add quite a bit of body to real signal that a simple EQ can't. A guitar speaker is naturally non-linear once you start getting 'loud'. The magnet starts to saturate, and the cone starts to breakup and gets harder to 'push' as the sound gets louder and the speaker reaches its physical limits. This isn't just the Fletcher-Munson curve that you're hearing (which is also non-linear), but the actual physical effects of the speaker/cabinet running outside of its linear response. So if you like clean guitar sounds, where the speaker is still very linear, then an IR can capture 99.9% of whats going on in the real world. But, if you want great distorted tone (ie. the sound a real amp can only make when you crank it up loud), then modelling the speaker non-linearities becomes critical. Without these (ie. if you just use a linear IR), the final result won't quite sound completely 'real'. I assume the Helix amp models are also modelling the non-linearities of a tube amp's output transformer into a 'typical' reactive speaker load. But, the cabinet model sounds like it's missing much of the non-linear magic that a real speaker cabinet provides.

The sp/dif 'ground' isn't actually grounded on the Helix, it may or may not be galvanically isolated. And may or may not be able to deal with significant ground differentials. It is important that both pieces of equipment are otherwise grounded. Have you tried using the sp/dif while the analog lines are connected (the XLR is best as it's definitely directly connected to the chassis ground inside the Helix)? Intermittent clock issues are incredibly rare with sp/dif, not impossible if the design has been botched, but unlikely unless the engineers have gone out of their way to fail. And 'turning on a fluorescent light' or 'touching a cat' is all about external noise being injected into the signal line. Also, clock drift issues tend to manifest with periodic orders of seconds or quicker, not minutes, and are usually quite regular. It'd be nice to see a proper circuit block diagram or a full schematic of Helix's external interfaces, as that would make it much easier to come up with proper solutions to all these interconnect related issues. The circuit topology of the output makes a huge difference to the solution, and not knowing means wasting a lot of time trying to determine the underlying issue (even if the Helix isn't actually the device at fault). Unfortunately, I've never seen any detailed circuit info come out of Line6 for any of their products, so it's unlikely they'll start now. Sadly, it is rare these days to find a tech company that provides detailed technical service/repair manuals for their products.

The HD500 and 500X used an ungrounded switch mode plug pack. The Helix has a proper earth connection through the IEC power connector. If you connect using the analog interconnects do you hear ground noise? When you say turning on a fluorescent light causes a dropout (and someone mentioned touching their cat has a similar effect), it really does point to some kind of grounding problem (or other hardware signal related issue). The hard part about diagnosing ground 'loops' in digital interfaces is that you can't 'hear' hum/buzz in the conventional sense. The issue manifests indirectly as dropouts, ticks, and loss of sync. Solutions are similar to solving analog ground problems. (Note, if you use a transformer isolator it must be designed for digital as the bandwidth is much greater than analog baseband audio.) Also, make sure you're using 75 Ohm coax for the sp/dif. Depending on the cable length, circuit design, termination, and signal levels, just using a standard audio cable (or 50 Ohm coax) may cause a marginal connection to degrade to the point where it starts to drop out. This really doesn't sound like a firmware/software issue.

Something worth considering is a ground loop problem. With analog you can hear the hum/buzz and can tell immediately that something is wrong. With SP/DIF it may seem fine, but occasionally drop out. Ground issues arise due to either an improper ground or an alternative ground. One simple thing to try is to use an XLR or 1/4" cable to connect the two devices (while also connecting the sp/dif cable). This should provide a good ground. No need to actually use that extra line for audio, just have the physical connection in place. Then check whether the problem still exists on the digital sp/dif connection. If the digital signal is still dropping then you can also listen to the analog line for ground noises if you suspect a typical ground loop. If the analog line is quiet AND the sp/dif is still dropping, then the digital interconnect cable might still be at issue.

A good idea in theory, however, the effort required to do this is huge. The maximum theoretical power transfer for an ideal P48 phantom power supply is 170mW. But, that would require an incredibly complex switch mode regulator design in the G series receiver to come anywhere near being able to draw that full amount. The G series digital receivers probably need at least 5 times that (and probably significantly more). It is theoretically possible, but would require a significant R&D effort and complete redesign of the G series wireless receiver most likely using a custom designed low power ASIC and phantom power regulator. It would be FAR easier to just include a G series receiver inside the Helix (or a special purpose power+signal connector that could connect one or more external G series 'modules'). For future designs of hardware like the Helix, I'd like to see more use of PoE (Power over Ethernet) which can provide considerably more power than P48 phantom, and is inherently a digital interface eliminating an extra AtoD / DtoA conversion stage between the interconnected devices.

When I installed them into a tun-o-matic bridge on a Schecter, I drilled a 3/8" hole under the bridge and met it with a hole into the bridge pickup cavity. Ran the 6 wires through the 'tunnel' into the bridge pickup cavity and then into the control cavity. If the control cavity had been closer, I would have skipped going into the bridge pickup area first, and drilled the intercepting hole from the control cavity. Completely invisible install that looked like it came out of the factory that way.