Posts posted by gckelloch
I'm still on WinXP. There are settings you should make to your system so the audio runs smoothly with external A/I devices. Look into that. It is recommended in WinXP to disable the MB audio when using another A/I device. If the MB audio is active while using the Pod through USB it may cause freezing or stuttering. At the very least, make sure any app assigned to the MB audio is not running when you plug in the Pod. Then, assign the Win7 Sound CP to the Pod. Not sure about W7, but I think you still need to stop the USB device before unplugging it, or you'll need to reboot before plugging it back in.
Hey, that could be that effect that occurred with the old Celestion (Greenbacks?) where a fifth below the fundamental note formed as the speaker cone was pushed beyond it's limits. Try using one of the pitch effects after a fuzzy amp tone set down a fifth mixed in with the amp tone. Probably better if it's monophonic so it's more squirly. Maybe even put it on a separate fuzzy amp tone path with the Auto-Vol. Echo or a triggered filter FX run before the amp set to trigger and mangle the sound a little.
I wouldn't count on any blond era models, but you never know. I'm no expert, but my guess is the amp section may not be much different, other than maybe the output transformers and/or PSU -- a similar response of which could be tinkered with using the DEP amp Sag. I'd think turning 6L6's or KT66's into 5881's using the Bias & BiasX might be the closest match, so there are a few options there. Unfortunately, you can't alter the preamp section, but some gain pedal placed in front with the drive turned down low might give you the type of sound you want there?
Have you tried the Divide 9/15 6V6-based clean channel (first drive knob)? The 5879 preamp tubes in the that model are supposed to have a fatter sound than most preamp tubes. Maybe...leave the amp DEP Master maxed like a real amp (and so you can actually hear the DEP knob changes); then turn down the Bias X for more crispness and/or turn the Bias up for more aggression. Adjust the Sag to taste. The PhD Motorway might be another option if the preamp section doesn't produce too much gain for you. If not, I think you'd just need to tighten up those KT66's like with the 6V6's, and adjust the Sag to taste again.
In other threads, we've discussed how some the preamps in the amp models respond to different input levels before the drive knob. You can try driving amps with an eq model for a more subtle or crisp attack quality. Whether it's different than adjusting the amp drive knob depends on where in the amp model's preamp section the drive knob is applied. If it in the first gain stage, there won't be a difference.
I think it should be made clear that it's the models you are using that cause the compression, and not something applied to the internal signal path regardless of the model. Otherwise, the same thing would occur when just using the mixer block. Yes, I imagine any model with a gain or drive knob would compress the input gain. That should come as no surprise.
In experimenting with on-line A/V meetings using a Pod HD as the A/I in a Mac system, I did notice that the A/V meeting app caused the Pod USB input to auto adjust as I spoke. That's because the A/V meeting app has built in auto-leveling. If you want to avoid that situation, make sure no other active app is sharing the Pod USB driver when using the Pod through USB.
Speaker testing is normally done in an anechoic chamber. Since they state -10db for the freq response in their data sheet, I think it's safe to assume it's at those top and bottom stated frequencies. It's also much more in line with the standards for bass response of modern speakers of that size. You may have a low frequency peak in your room where you measured the response. It could be very different somewhere else in the room. There can be cancellations at node points on the order of -20dB in untreated rooms. As you said, monitoring position is important to experiment with.
The other thing to consider when monitoring with headphones is the lack of speaker "cross-talk" to opposite ears that gives stereo speakers a particular sound stage. The stereo image will normally be very different through either medium, but there are computer plug-ins and dedicated preamps that simulate a stereo speaker image. That's all I can really suggest on this matter. I hope the ideas presented offer you some guidance in choosing what works best for your needs.
Sansui? Just vaguely. I recall it being on the level of Nakamichi. The OP mentions his Denon receiver -- great stuff as well.
In further investigation, the 'Rokit Cut Sheet" lists the FR specs at -10dB points -- just what I was getting at. There are no FR graphs I could find. The Equator D5's initially gave FR graphs on the site, but I can't find them now. I do recall the FR graphs being at -3dB point as listed in the specs. Why they are no longer accessible seems suspect to me. The RokitG3's might be tuned to accentuate thump/punch at ~100Hz, so they may be more of what the OP is after in that sense. No reason he couldn't save money and still make use of that great old Denon receiver with some decent passive time/phase aligned speakers.
Good points, RIblues. I think between our postings, we illuminate more things for the OP to consider in his/her decision. The Rokits may be more to the OP's liking...and they may well have more exaggerated bass, but not necessarily lower bass extension. It's actually the case that sealed cabs generally have more accurate and more extended bass -- particularly in average-sized bedrooms. There is certainly a case for matching your rig closer to average consumer listening rigs with typically hyped bass and highs.
Well-balanced speakers with phase/time-aligned Xovers can seem to favor midrange because there is much less frequency waver in the Xover range, but the reality is they are just flatter/clearer in that Xover range and should be closer to the sound heard through headphones. We aren't generally accustomed to hearing that in home speaker systems, but professional active Xover PA systems are now often time/phase aligned. I think the answer to getting a consistent and full sound between patches while minimizing disturbance to neighbors is close-field monitoring. You can't expect to get that physical kick in the pants bass without the rest of the house hearing/feeling it, but you can get that kick in the chest bass with close-field monitoring.
Regarding the variable loudness feature I alluded to: it was first offered in Nakamichi Receivers in the 80's to set the proper bass and treble loudness range boost in relation to the volume level. It was connected with the volume knob, so it could be set by the user after factoring in the variables in a given listening environment. All audio reproduction systems should really include such a feature to equalize the loudness curve regardless of the volume knob setting.
You should never put monitor directly on the floor or a larger flat surface. The bass response and early reflections will completely alter the sound -- bad if you are trying to set up presets for FOH pa systems, or any well calibrated audio system for that matter. You best bet is too actually stick with a close-field 2-3ft. monitoring situation, because your room resonances and reflections will not significantly color the sound, and you get better results when tweaking sounds.
The thing regarding powered monitors is that there is no loudness compensation like on a typical Hi-Fi amp/receiver, so the bass and treble would not be as loud as the mids when the volume is down lower. For that matter, a home audio receiver with variable loudness compensation and some fairly flat sounding low ear-fatigue (usually with a silk-dome tweeter) bookshelf speakers might be better for you? I had recommended Warfdale speakers for that scenario, but there are probably some other great options for that purpose -- preferably of a coaxial design with time/phase-aligned Xovers. The single driver Avantone MixCubes are actually a poplar option -- probably great for guitar. Otherwise, powered monitors of such design and a quality preamp with a variable loudness knob wired before them might work better.
If choosing the latter option, some of the afore mentioned monitors seem good, but close-monitoring really requires the drivers to be very closely spaced, or ideally coaxially mounted for a signal point source. I've recommended the ~$400 Equator D5's in other threads. They are a coaxial silk-dome tweeter monitor with other features like time/phase-alignment, and DSP driver frequency correction for the accuracy crucial for getting the critical upper-mids in your guitar presets right. They have a -3dB point at 53Hz, which means the lowest note on the guitar will be accurately represented. The next model up is the D8's with a -3dB point at 45Hz, which is a little better for full range music. They are $750 though.
If not specified, the bass cutoff point for many speakers is somewhat deceptively given for the -6dB point. Kevlar does not necessarily create a less resonant structure than mineral-filled poly. Kevlar is just a stronger material. It will not create a completely rigid structure, so some resonance will occur. The resonance, while being slightly lower in level, would be higher in frequency -- possibly muddying the crucial vocal midrange. Low frequency resonances are much less intrusive to overall clarity. Mineral-filling is really more effective in reducing resonance, as well as lowering the bass cut-off point for a given speaker-size due to the increased mass. All factors in design must be considered for a certain result. If any one factor is substandard, those intended results will be compromised. FI, the cabs and ports of the Beringer 3031A's are known to produce audible resonances up close, thereby nulling any supposed advantage of a Kevlar woofer.
It's common marketing practice to highlight one "feature" as being superior for unrelated reasons, or to present very common "features" as being unique or innovative. Vagueness is often used to imply things that aren't really correct, or to create confusion in hopes that potential customers will simply "give up" and "give in" to emotionally-based hype -- all manipulative tactics we can thank consulting "sell-out" psychologists for on Madison Avenue.
MartinDorr, incorporating a soft-clipping feature would hardly be worth keeping a secret. Nothing new there. I doubt there is one, though. However, it's not uncommon to calibrate the D/A converter for 6dB analog headroom before 0dB, which could mean analog gain compression starts at -6dB. I used to accidentally drive my old EMU 0404 soundcard to soft-clipping. I've not seen it ever calibrated to -12dB though.
The meters in Reaper would register any clipping down to 1 sample. You also said that some models don't seem to have any gain compression even right up to 0dBFs, but then warned against driving the internal signal that high to avoid any gain compression "...that does not come from the modeled gear but internal HD signal level limits". That could not happen if some models don't compress up to 0dBFS.
If the Pod HD USB driver is calibrated so when set up all the way without the 18dB boost engaged to be the same as the Pod HD D/A, and your software meters are set the same, then you will have a reference to judge clipping by. I suspect the USB driver is calibrated as such, so you might test if internally driving the Pod HD USB output past 0dBfs in your software causes clipping through your analog outputs. Again, those outputs might have 6dB or more headroom for good measure, but it would be worth getting straight.
I got the message. Good vibes to all those involved. It's all about connecting. Let that be the driving force.
I'm curious what you got. Can you post a link?
Sorry if what I wrote was overwhelming. The important thing to be aware of is how the Xover in your speaker affects the response in the overlapping range. Some passive Xover designs have flatter response than others. You can probably get some passive speakers that are very good in that respect for under $200. No reason you have to get active speakers if you have a decent amp. This article shows the response of some common passive Xover designs: http://sound.westhost.com/ptd.htm#s2
The response of the design in fig. 8 would be best for your application, because the variance is small, and a slight compensating boost around 1.5kHz on your part would not be hard on the audience's ears like a larger boost between 2.5-3kHz would be. So, check out the Xover frequency point and design of whatever passive speakers you consider. Maybe look into some Warfedales. Something with the drivers aligned to the same distance from the listener might be worth considering as well. My brother got a decent old pair of 2-way JBL home listening speakers with an 8" woofer and silk dome tweeter for $5 at a yard sale. They aren't good for close-monitoring because of the distance between the drivers, but they are fantastic for what he paid for them.
Whatever you get, you'll want to elevate them off the surface of your desk up to ear level to minimize treble slurring reflections and bass/mid coloring. Keep them several inches away from the back wall as well. If you set them on either side of a PC screen, I'd place them so the inside edge of the cabinets are flush with the screen surface. I also drape a thick towel over my desk when mixing. If there is a wall right behind your head, you should probably hang something on it to damp hi frequency reflections as well. Acoustically treating your room is usually not necessary for close-field monitoring.
For roughly the money you mentioned, you could get some great headphones, and maybe a monitor simulation headphone amp. There are computer plugins that simulate monitors as well. You can get quite accurate mixes that translate very well to real speakers. If you are set on inexpensive passive monitors, consider the Tannoy Reveals for comparison. I've also heard that Warfdale makes some great inexpensive passive monitors. Otherwise, read on.
The HS5's were the monitors to beat in the low price range. They may incorporate phase-aligned crossovers and/or be time-aligned. Yamaha does not specify, so I doubt it. They're also in the same price range as the Equator D5's. The DSP correction essentially puts the D5's in a next generation category. They also have a 60-day money back policy. The price just went up to $400/pair...my bad. Check out the on-line reviews in Mix, SOS and EM Magazines.
Crossover overlap distortion is an issue in multiple speaker enclosures. D5's don't have more midrange. They have clearer midrange than other speakers in the ~1-3kHz region due to the phase-aligned crossover and the physical time and position alignment. 1-3kHz is mostly within the critical hearing range (2-4kHz), and is the most significant region concerning guitar tone definition -- as well as for many other vocal-range instruments. If something is off in that region, the sound can easily be harsh, edgy, "phasey", or just ill-defined. Guitar amp tone stack midrange is centered around the ~500Hz region, so it's not the same thing at all. The D5 single point source coaxial design is also better for stereo image accuracy with a wider sweet spot in a close-field setup.
Ted Keffalo (formally of KEF and Event), is a major mover in the audio world...and a master of all time and space (pun intended). ;) I have the Event ALP5 time/phase-aligned 5" monitors -- the last model Ted designed when at Event. The highs are essentially flat/even and sweet. Bass response is outstanding for the size: -3dB at 53Hz, just like the D5's -- more typical of an 8" monitor. The highs and bass are the more obvious aspects people hear when comparing monitors. The D5's reportedly have non-colored highs provided by a non ear-fatiguing silk dome tweeter. They also have the lowest in class bass-extension -- probably just like my ALP5's, but clearer due to the DSP correction.
No speaker is near as flat as manufacturer FR graphs would indicate. The response is commonly smoothed out to 1/3 octave -- meaning the many variances occurring inside those 1/3 octaves are not represented. The only way to compensate for those variances is with DSP correction specifically designed for the speaker. DSP filtering has developed to the point in recent years of being virtually transparent with no audible artifacts.
I know of no powered monitors in that price range that produce accurate results with the clear midrange crucial for getting your presets to sound correct recorded, and through quality full range systems. Upgrading to inaccurate monitors is just not worth the money. Phase/time alignment and DSP frequency compensation is really the way to go. At ~$380 shipped, the Equator D5's are the best bang for buck going. Although they have the option, balanced XLR's are not necessary for short cable runs.
It can't really be done without a Variax, but one of the pitch shifters set an octave up and slightly detuned should approximate it accept for the top two strings. I think Howe used 60's Marshalls, so the JTM or Plexi lead amp might be best, but a Vox or the Dr. Z might be really cool.
I wonder if it also changes the output impedance? It would make sense if it did. Not sure if it needs to though.
Amen to that! I think they presently underestimate the value of customer interaction and feedback. Perhaps they should derive philosophical analogies to customer relations from the functioning of the equipment they model?
Well, once the input signal hits the amp/s it won't really affect the output. As to whether it may clip FX before the amps is a matter of debate. My guess is that some of the older Pod FX are 20 to 16bit resolution and may clip more easily. It's kind of lame for them not to have documented this stuff so we'd know for sure.
Impulses for speaker cabs? Yeah, they might need more latency not to max the CPU.
Cab impulses are normally limited regarding response/feel. It depends on the plugin. You should consider some kind of preamp and/or tape sim plugin to run after the cabs to tame resonance peaks. Otherwise, a "brickwall" limiter would suffice. Compressors won't catch the initial transients. I prefer natural gain compression -- Rupert Neve/Eddy Krammer old school styley.
Really nice free VST dynamics plugins here: http://varietyofsound.wordpress.com/vst-effects/
I like running the TeslaPro transformer saturater into the FerricTDS tape saturater -- no limiters needed.
Is your CPU maxing out?
You using the actual ASIO drivers and not the ASIO DirectX or Multimedia driver?
You sure your ASIO driver buffers aren't set too low? If not, reset Reaper to default settings form the Reaper start programs folder, or install any other app that uses ASIO. Reboot and see if it works.
Maybe your Reaper disk stream settings got changed, or you have a failing disk?
Absolutely, Deano. -18dB is the pro digital standard recording level. it still leaves 21 bits out of 24 for a theoretical noise floor of 126dB -- more than ample resolution for accurate low level low frequency signals. I've run tests with sine waves recorded in 16bits and 24bits. Generated frequencies right below -70dB in the ~300hz - 1kHz range all become quite deformed, creating harmonics and weird modulations -- while the same affect doesn't occur in 24bits until well below -100dB. Most pop music is around -12dB RMS, so the low frequency wave deformations are then ~60dB below average. That's pretty low compared to the average level, but I think we can hear the affect with dynamic acoustic music as slightly muddled upper bass/mid harmonics during quite passages. We've been listening to CDs like this for decades.
Sounds correct. It should only be 6dB hotter if both paths have identical signals and are panned center. Some dual amp configurations will be louder than others when center panned.
BTW, the Tascam mixer might just be 16bit through USB. Pod HD's are 24bit, so you'd be loosing some resolution recording with the Tascam.
I can't say whether engaging +18 in the Pod HD USB driver record level actually matches the internal Pod HD signal level or not. My guess is: with the USB driver record level up all the way and the +18 off, if the dry/direct fully side-panned guitar signal going through the Pod HD produces peaks in your DAW up to anywhere from roughly -18 to -6dB when strumming really hard, then that is probably the same level of internal headroom inside the Pod HD. I'd use that dry guitar level as a reference for peaks for your FX presets. If those dry signal guitar peaks are down near -18dB (pro digital standard recording level), then you should be able to push the Pod HD internal signal up another ~12dB with ~6dB headroom before internal clipping, and you certainly wouldn't need the USB driver +18dB boost. That's my guess, and it's the best answer I can give because Line 6 is not specific about the internal levels.
Fellow Reaper user here -- no better DAW choice out there IMHO. Recording with peaks between ~-12 to -6dBfs in your DAW is good. The pro standard is actually -18dBfs for excess headroom for send FX. That's still over 120dB resolution with a 24bit A/I, which the Pod HD's are.
Section 9 in the "POD HD500X Advanced Guide - English ( Rev A )" tells you how to adjust the guitar monitoring level. The 300HD must have the same info. You should get best quality recordings if you record at 24bit 48kHz. Realtime SR conversion while recording is inferior.
Yes, you want to maximize the output signal without clipping. I've already explained how you might loose L or R signal content with stereo FX on either of the dual paths. You were correct. If doubling a mono signal by centering both pans does not clip anything after the Mixer, then that's fine. I depends on how high the FX outputs before it are anyway. Just lower the Mixer levels if needed. You got it figured out now, chief. ;) You're good to go.
Use external metering to make sure you never exceed 0dBfs thoughout the signal path. Check to make sure your peaks are between ~-12 to -6dBfs. If a few peaks go a little over -6dBfs when you strum hard up high on the neck, you will be using the full bit resolution and maximizing S/N. I wouldn't let peaks go any higher than that though. Any modulation FX run after amps can create wild peaks. Either use a DAW meter through USB, or use a phone SPL app with some monitors. Assuming it isn't clipping the input, use the direct/dry/unprocessed guitar signal as a reference. If the direct guitar signal peaks don't ever go much over -12dBfs, that's normal. You can set the processed signal/s up a little higher. I'm fading, so I better be off for now. Good luck.
Question About Using The Pod Hd With A Daw
in POD HD
Wait now...as long as you don't use any DAW FX that introduce latency, you should be able to get sub 10ms latency through a DAW. The manual shows a buffer of 256 samples in the PC USB driver, and says it can be set lower. I set my Prosonus Firebox to 6ms (256 buffer samples) and it never throws me off. There is also a ~1ms added delay for every foot you are from your monitors. You should record at 24bit 48kHz SR -- as it's the Pod HD native rate. You can cut the latency as much in half by recording at 96kHz, but you won't gain any signal resolution and the total CPU use will double. At other SR settings, the real time USB driver SR converter may add latency. Real time SR conversion is inferior and could possibly cause a midi timing sync issue that builds up over time -- as was the case with the old SBLive soundcards.