Meiannatee

Let’s not start pointless unrelated arguments that benefit no one. First, Pianoguyy, do you mean that if I live in an apartment (like 90% of the population in Singapore), can only rehearse in a small studio, and have 15 minutes to sound-check at a live venue where I’ve never played before, then too bad, bro? We get your point, but those of us whose situation is further from ideal, we want solutions, given our current situation. Thanks for informing us of our situation, because awareness of our problem is the first step. Rodmoyes, if you don’t mind, I’ll try to explain your point. Let’s say I’m going direct into the house PA. I set the master volume on the Pod to 50%, level all my patches, and save. Now, I have consistent loudness across my patches. Then, I select a patch, increase my master volume to 100%, and have the sound guy compensate by lowering the guitar fader, such that the loudness of this “calibration†patch is exactly the same before and after. Now, if I change patches, I would expect them to be exactly as I levelled them before, but in Rod’s case, this is not true. With loudness perception taken out of the equation, the master volume control changes the volume balance between patches. Some explanations I can think of are: - The master volume control interacts with the volumes of FX blocks in an undocumented manner - The output level is too hot for something downstream of the volume control at 100% (DA converter?), resulting in compression/limiting; sounds with lots of bass gets compressed more, resulting in uneven levels Rod, please don’t be so quick to sell your Pod. Try my method or someone else’s for a few more months? If you try levelling each FX block like mine, of course just increase your amp volume to compensate for lower input volume. If it doesn’t work, at least you tried. Now, if you wanna talk math, I’m no expert, but there seems to be some confusion between sound pressure level and sound power level. Sound Power Level This is the rate of emission of acoustical energy. It is independent of room and distance. Sound Pressure Level This is the variations of pressure above and below atmospheric pressure. Sound pressure level at a particular location depends on: - Sound power level of source - Directivity of source - Distance from source - Room volume - Room surfaces and furnishings When someone refers to “bedroom levelsâ€, it refers to the sound POWER level of a source (amp, monitors, headphones). Conversely, the Fletcher-Munson curves deal with sound PRESSURE levels. Sources with different sound power levels can create the same sound pressure level in a listener’s ear, depending on the variables above. Example: Consider a 90dB Sound Power Lvl (re 1 picowatt) “bedroom amp†source, compared to a 120dB Sound Power Lvl house loudspeaker. 120dB Sound Power Lvl is 1000 times more powerful than 90dB. For simplicity’s sake, assume a venue with no walls,100% absorbent floor, identical directivity (hemispherical/conical radiation), and no speaker distortion. A listener’s ear 1 meter away from the “bedroom amp†will experience the same sound PRESSURE level as another listener 31.62m away from the powerful house loudspeaker. In these scenarios, loudness perception across the frequency range is identical. If we place the bedroom amp in an untreated room and the loudspeaker in an outdoor arena, not changing the distance variable, the bedroom amp will sound even louder, due to room reflections. Bonus content: A thought experiment I estimate a pair of really loud headphones to be operating at 20-30dB sound power lvl (re 1 picowatt), which is around the sound power level of rustling leaves and a soft whisper. This puts the headphones 1-10 million times less powerful than the 90dB bedroom amp. How then can the headphones make a listener deaf with prolonged exposure, but not the bedroom amp? My Answer: A combination of factors: - The source is over 100 times closer to the ear. - The “room†is much smaller. This room being the air space in the headphone cup and ear canal. - Greater directivity, since the acoustical energy has not many other places to go. These and possibly other factors result in a sound pressure level across the ear drum which is greater than the danger threshold, thereby causing noise-induced deafness. Again, I’m no expert, so please correct me if I’m wrong.

First of all, I don’t claim to be an expert. Just sharing my experience. I think that the OP’s argument is valid, since not everyone has the luxury of creating their patches at gig volume. However, I don’t struggle as much with volume levelling and master volume control since I try to level each effect block such that the on and bypassed volumes match. Yes, even amp blocks! Most users will say that I’m doing it wrong; that an amp output should be much higher than its input. Well, if you’re a heavy effects user, you may want an otherworldly ambient tone without an amp block. Then you’re forced to use the mixer or one of the fx’s gain control to match the volume of your patches with amps. This just complicates things, and I like each block to be as modular as possible, i.e. I can swap them around without worrying about volume levelling. So, I start with all effects levelled, then tweak a little bit, like maybe giving the appropriate fx blocks a slight boost, or raising level of the overdriven amp blocks a little above clean ones. Nothing major. So far, I get fairly consistent levels at various master volume settings. Am I doing it wrong? Not sure, but it’s working so far. I think of it as something like mixing, where we mix at lower levels (-12 to -24dB) with plenty of headroom, and leave the loudness part for mastering. Well, I’m processing my tone with plenty of headroom, then using the master volume control to match whatever’s downstream. Of course this method does not apply for those who use the Pod solely for effects and use a physical amp for distortion.

I suppose you mean adding a pitch effect in front to transpose down an octave? No, I tried that and it doesn't work. It works as a makeshift bass for jamming with the looper, but it definitely sounds like a pitch-shifted electric guitar, not a bass guitar. I got a cheap bass for that very reason. To me, a $200-300 Ibanez/Yamaha/Cort will sound better than any pitch-shifted guitar. Try the used market as well; you may get a good deal depending on where you live.

Good point. However, I only go direct or headphones, so this probably doesn't apply to me. At least for me, this level makes it easier to balance, say, patches with no amps, or with an amp in only one signal path. Some of the spacey tones work better without an amp model.. Then you'll have to waste an FX block to boost your signal, just to level that patch. 50% Master out gives me a comfortable headphone volume, and around 90% puts the mixer I use on the verge of clipping. If you're going into an amp, do listen to edstar's advice. But even so, don't make your patches too loud, since you'll want the amp to be pretty clean if you're using delays and reverbs. Or else, the post-delay/reverb distortion can cause your sound to be washy and get lost in the mix.

Seems like you're crafting an amp-less patch, maybe synth or soundscape? I can think of a few... Mid focus EQ, high-pass 0%, low-pass 100%, adjust output for clean boost. Parametric EQ, default settings, adjust output. Vintage pre-amp Tube comp Boost comp Also, check that both channels in the mixer are on, and panned center. But for me, I think that perhaps the problem is that the amp patches are too loud, especially the presets. I level my patches to match the loudness of a blank patch. I'm probably the only one who does this, since Customtone patches are crazy loud as well... Be careful as you can clip your signal if it's going in too hot into certain effects.

Good points, and here’s how I’ve been trying out the amps. 1) Make your own “blank†preset. Mine’s not completely blank. I have a hard gate as the first FX, and a very gentle room reverb as the last (you may need a reverb if your room is very dry or you are using headphones). Rename the preset (BLANK, EMPTY, anything you like) and save. 2) Copy your “blank†preset to an empty slot. On PC it’s Ctrl+drag. Add an amp and cab of choice, do a rough adjustment on their parameters. Usually I like to reduce the Master DEP to under 60% to remove “cross-over distortionâ€, as some forum members here have pointed out. 3) Add a mid focus EQ. I set my high-pass to around 10-20% to remove boominess, and low-pass to around 70-85% to cut fizz. Other EQs like the parametric works too, but at this stage, the mid focus is faster to work with. 4) Level the patch. As for me, I use my “blank patch†as a reference, to get roughly consistent loudness. Because of this, my patches are usually quieter than Customtone ones, but it works for me. Note that some effects, like the EQs, can digitally clip if you send too hot a signal into them. 5) Save. Copy this patch to another slot, change the amp and/or cab, roughly dial in amp, cab and mid focus EQ. Save and repeat. This way you can quickly come up with a handful of your favorite amp-cab combinations. Now, you can fine-tune them if you wish, then back-up on the computer! Using these as templates, copy them and add your desired toppings. This is just my way of doing it. Beats starting from scratch every time. I’m sure others here have different ways of doing it that works for them. (Additional note: I put a hardgate/noisegate at the front because I'm using the settings Input 1: Guitar, Input 2: Variax. The gate sums the left and right channels to mono)

Added graphs to show just how simplified this example is. The details can be found here: http://en.wikipedia.org/wiki/Loudness As you can see, it's way more complicated than the smooth curves and straight lines in my example, and the pattern differs across the frequency spectrum.

Actually, if you’re talking about sound pressure, if the taper is linear, you’ll have the opposite effect to what you’ve described; at heel position, changes would be drastic, while at toe position, almost nothing happens. Let’s look at a simple example. (All log have base 10). Let x be the volume pedal position (0 = heel, 1 = toe). Let y be the sound pressure ratio [p/(reference p)], such that log y = 0 is the minimum volume, and log y = 1 is the maximum volume. When x=0, log y = 0; when x=1, log y = 1 For an audio (logarithmic) taper: y = 10^x log y = x For a linear taper: y = 9x + 1 log y = log (9x + 1) As you can see, both equations are calibrated to have the same min & max levels. Since the human ear perceives loudness in a more or less logarithmic manner (for simplicity’s sake, I’ll not go into the details), the slope of the audio taper should be (more or less) constant in a plot of <log y> against <x>. Audio taper slope: log y = x Let Y = log y: dY/dx = 1 Linear taper slope: log y = log (9x + 1) Let Y = log y: dY/dx = (log e) (9 / (9x + 1)) = 3.909 / (9x + 1) Solving this equation of slope of the curve for x = 0: dY/dx = 3.909 Solving for x = 1: dY/dx = 0.3909 Solving for dY/dx = 1: x = 0.323 Conclusion: 1) In this example, initially when in heel position, sound pressure level (SPL) will be 3.909 times more sensitive to pedal position for the linear taper, compared to the audio taper. Recall that the slope of the audio taper curve is 1 (constant). 2) This sensitivity drops hyperbolically when the pedal is pushed down. When the pedal is at position 0.323 (32.3%), linear taper and audio taper has equal sensitivity. After this position, linear taper has lower sensitivity. 3) At maximum toe-down position, linear taper is 0.3909 times as sensitive as audio taper (or 2.56 times less sensitive). From my experience, it seems like it’s pretty close to, if not exactly an audio (logarithmic) taper. Please correct me if there’s anything wrong with my lunchbreak math ;) Edit: http://www.docdroid.net/ywak/sound-pressure-level.pdf.html Graphs are worth a thousand equations =)

This can be fixed if they would allow the pedal to control mixer level. Currently, to free up an extra block, you can set the pedal to control amp volume, and adjust max volume either by changing pedal toe-down value, or adjusting the mixer. Mildly annoying. Another annoying thing is that I can't resize the window on the Edit software. Things are really small and hard to see on a high-res monitor. I imagine it would be horrible for people with 4k monitors.

I used it on Easter too! :lol:

As mentioned in the title, I humbly seek ye veteran “Pod heads†for advice on dialing in a tone specifically for double tracking. Looking for something like Tesseract’s rhythm tone. I think they are using 6505s. I would be helpful if someone can suggest the best amp/cab/mic combo for this, and the settings on them. I’ve been messing with the 5150 (PV Panama) with a Screamer and hard gate before the amp, and mid-focus EQ after the amp. Sounds pretty good when playing alone, but double-tracked in a mix (plus drums and bass), it’s maybe 70% of a solid rhythm tone; sounds good, but maybe lacking a bit of punch, or a little something that I can't put my finger on. From what I gathered, for double-tracked guitar parts, I should lower the gain a little to increase clarity. My Mid on the amp is currently quite high at around 80%. Exactly how different are double-tracking tones compared to single ones, and how differently should I approach this? Is there a recommended process for dialing in tones specifically for this? Right now the only reliable way I can hear the end result is to record the sound double-tracked and listen to the playback. Quite a long, incremental process. I don’t really play metal, but need this for my latest project. And it feels like a great opportunity to learn something different. So, any help will be greatly appreciated.

Sorry I meant "not too wide". You're right, higher Q% = narrower

Actually, I was wondering about this: Let's say I have fizz at around 3kHz, like in Peter's video. 1) If I use the mid-focus' low-pass filter, I'll be attenuating at around 3 kHz AND everything above 3kHz. 2) If I use the parametric, if Q is not too low (wide range), I'll only be attenuating frequencies around 3kHz, while leaving the higher frequencies alone. 3) If I use both, I'll perhaps use the parametric mentioned in point 2 above, while the mid-focus gently rolls off the higher frequencies, maybe 4.5-6kHz? Since I'm such a noob, some days I find one method sounds better, some days it's the other one. So, there's really no difference?

Thanks Peter for the effort in making the videos!! Really helpful. Hmm and since scias mentioned swells, I was thinking if it is possible to do warm pads/cello/strings-like sounds on the Pod (like the EHX stuff). I've been experimenting, but only managed to get a synth lead tone. Really enjoyed your playing, and will be looking forward to any new video! by the way, where can I find your patches? Thanks! Edit: Nevermind, found the patches on your website, thanks! Was searching in Customtone and couldn't find them.

Very useful video!! Any idea which would be better: using the parametric EQ, mid focus EQ, or both? So far I've been using the mid focus by itself to cut fizz. How about you guys? Wow, you must really love your Pod if you're making out with it! :wub:

Yea, there's a lot of good stuff in the model packs. Bought them all and not gonna return them! (No matter what the wife says..) Besides a full (modern) studio compressor, there's the EQ thing, like Hurghanico mentioned. Like, who uses the bass and treble controls on the parametric and mid focus EQs??? Why not add more bands to the parametric instead? And I wouldn't have such a big issue with the frequencies being shown in %, if it wasn't for the fact that when i change EQs, the same percentage value corresponds to a different frequency! ARRRGH But that's for another thread.

It's a shame that they didn't make a model of just ONE compressor with the 5 controls mentioned, as this single compressor would have done the job in most situations without the need to change the compressor model! Yes, it may take more work, but hey, it's a Pod, and I thought Pod owners love tweaking :P Taking your advice, I think I'll make my patches after I select a comp so that I can EQ with the comp turned on in the chain. Then I'll just leave it on, and if I wanna turn it off, I can make another patch without the comp and with different EQ settings. The application in which I need compression is using a piezo pickup in a live setting (FRFR). Those buggers are annoyingly overly dynamic, and the sound guys complain about the plucked stuff being lost and the strummed stuff being too loud. This is also the situation where the tone effects of compression is not wanted at all. I don't need compression on my dirty tones, and even clean ones with mag pickups. Regarding compression and mixing, sound engineers actually like to have light compression during the recording process. Then, a little bit more during mixing/mastering. Gentle compression in multiple stages sound more natural than a one-time heavier compression. And if you're talking about "the good old days", music is already being compressed during recording, since everything's recorded on tape, and you get tape compression/saturation, if I'm not wrong.

Thanks for the input. I understand that the Pod’s compressors work like compressor pedals. I’m coming from the experience of using a studio-style compressor with a complete set of controls, like those found in DAWs (the G3 has something like this). I’ll elaborate (sorry, quite lengthy): What I do is set a moderate ratio, maybe 4:1, set the threshold high, make-up gain at unity and attack and release at default. At this setting, I'm not working/pushing the compressor: I can turn the compressor on and off with no effect on tone at all! Then, as I gradually lower the threshold, I start to notice more and more compression in the form of “flatter†dynamics, but the changes are gentle and gradual. This is not the case with the Pod comps. Even at the absolute minimum levels of compression, there is a change in tone when I turn a comp on and off. It doesn't gradually come in; it's there as soon as you turn it on. Due to my level (or lack) of skill, it’s tricky to identify if the comp effect is adding an EQ, or what I’m hearing is the result of the comp shaping the transients. Then again, if there was a comp pedal with enough parameters (threshold, ratio, attack, release, make-up gain) I could fiddle around and learn the comp. In particular, attack and release, since like you mentioned, they can affect tone perception. Can't do that with only Level + (variations of threshold) to play with... Nevertheless, I’m motivated to mess around with the tube and vetta comps again, once I get home from work. :)

I've had the HD500x for almost a year, and I must say that it's very convenient to use live, going direct and using IEMs. 1) Compressor control parameters However, I've been struggling with using the compressors (and overdrive pedals, but that's another topic). Having used a Zoom G3, I find the compressors on the Pod quite lacking in tweakability. There isn't a single compressor where I can control the attack, release, ratio, and make-up gain. Being able to control mix would be nice too, but I guess that can be done, only post amp, using the 2 signal paths for parallel comp and using the mixer to balance. With so many parameters to tweak in the "HD amp models", it makes one wonder why there isn't a single tonally transparent compressor with controls for attack, release, ratio, make-up gain, and mix. The Zoom G3, with all its shortcomings, has this kinda comp, and perhaps I miss having that level of control, more than I would miss having control over hum or sag. 2) Transparency The other matter is the way the compressors color the tone, even at the lowest compression settings. This is quite noticeable when A/B-ing a patch with comp on and another with comp off after loudness equalization. Even the tube comp does this, although to a far lesser degree. It distorts (mildly) fairly easily at higher levels of compression, which may give that added warmth that some like, but messes up clean stuff like strummed parts with piezo pickups. Side note: the output gain control does not go low enough for equal loudness with the effect on and off. Even at 0, output is quite a bit louder with the effect on than it is when off. For these reasons, I've been avoiding using compressors. Perhaps they are due to my own ignorance/mistakes. Except for this and other small issues, I'm pretty happy with the Pod. Can anyone help me get around these issues? Thanks!

It could be because of the 20Hz high-pass and 20kHz low-pass filters on the global EQ in default settings. I'm no expert, but I've read here and there that people use low/high-pass filters to cut out those inaudible frequencies during the mixing/mastering processes to "clean" up their tracks. Yes, it may not make any noticeable difference most of the time, but sometimes those sounds could affect compression, excite (unwanted) harmonics in the audible range, and other stuff. So that's my explanation, but I would be totally fine if I'm wrong and someone has a better answer. Better tone is still better tone :)

Hi, I'm new to Line 6. Sorry to crash this thread, but I'm considering purchasing the HD500x. I agree that I have no right to demand for new updates if I already owned a HD500/HD500x and have been using it since its release. However, I'm considering purchasing a product that has been out for some time, and would appreciate an inside opinion on whether I should: 1) go ahead with the purchase because: a) the product's performance is not outdated and is competitive with other floor multifx/software solution in the market at that price range b) the product will be updated in the future to be competitive OR 2) I should wait because a successor to the HD500x is around the corner (2015?)? December sale is starting soon, but here in Singapore, the return policy is really bad, only covering manufacturing defects. So any advise I get will be really helpful. Thanks! Background info: I'm currently running a Zoom G3 thru an amp at church (super lightweight for public transport). But because of stage volume, micing difficulties and mic bleed, I'm considering the HD500x to go direct into PA.