pfsmith0

I've noticed no appreciable latency using the HD500 SPDIF output into my RME UFX interface.

Meiannatee, you forgot picks. They come 2nd in importance just behind strings.

This happened to me as well. All I did was plug in the USB cable slightly off-center and "snap", the black plastic center piece fell out.I was able to replace the connector without any issues at all, but I'm used to soldering and taking things apart (MSEE degree - old school, back when we actually touched circuits). I removed the PCB to get access to the top and bottom layers and had to suck solder off both sides of the old thru-hole USB connector to get it out without tearing up any traces. You don't need to replace the USB chip if all that's broken is the connector. A great USB connector replacement can be bought here from Mouser. In case the link doesn't work, the Mouser part # is 649-61729-1011BLF. They cost less than $1 USD each and are quite robust.

Plug your guitar straight into your Win10 Audio Interface and see what levels Audacity reads. That'll tell you how much gain you'll need from the HD500X. If it's a lot (perhaps 20dB or so) then you may need another interface or put in a few Studio EQs FX blocks with the proper gain selected.

Try the Studio EQ. It's clean and has a lot of boost capability.

Gosh, I never answered these! Sorry... I used an Audio Precision APx-525 to measure this. It's an excellent very high-end piece of audio test equipment. We have about 7 of these at work, each about $25k. I plugged the generator output into the HD500 (Guitar, aux, mic depending on what I wanted to measure) and routed the SPDIF back into the APx (unless I was measuring the output levels at the various connections). I did measure the clip levels of the various inputs and outputs here but it is hard to tell if the clip levels are due to the converters or the supporting analog circuitry. It appears a -30dBV guitar input gives a -39.3dBFS SPDIF level. This means the ADC will clip at +9.3dBV. But looking at the outputs, they all clip at 8.3Vpp, which is also 9.3dBV, So the clip levels seem to be due to the ADCs, not the DACs. I did not do any measurements on tonal changes as the input gets close to fullscale. X axis = frequency. Treble is toward the right. Bass is toward the left. Y axis = magnitude (aka amplitude aka loudness). The higher the graph the louder that frequency is. The main use for these particular curves is to give you a broad feel for what the various mics do. For example, the 57 will be somewhat bright because it has higher levels on the right side of the graphs. The 409 will give you great bass since it has a higher response toward the left.

This post may help. For example, you can see the 57 will be much brighter than the 421 or 409, but the 409 has extended bass response. Or you can see the ribbons offer more bass AND more treble. It provides a visual way of seeing what you'll be hearing. None of them are right or wrong. They just offer another way to shape the tone.

Hmmm, I have the older HD500 and use the SPDIF into and RME UFX which is setup as my Windows default sound card. I always have the USB connected to edit the signal chain and I always use the UFX to listen thru the monitors or headphones. I have never had a problem with both connected. I'm 80% sure this has to do with Windows sound settings. At least you know it IS possible! I would not use ASIO4ALL. Use the driver that Line Monkey provides.

Thanks bj. I wish I knew what BW they used to measure the noise floor since noise is dependent on measurement BW (whereas a sinewave is not). I recently moved and have not yet setup my DAW so it'll be a while to get anything from me. My technique, though, would be to only look at the SPDIF output to analyze the ADC/input circuitry. I don't I have the capability to accurately measure the DAC/output stages anymore as I lost access to the Audio Precision equipment. But my RME UFX ADCs "might" be good enough to measure the HD500 output. I'll have to play with that once I get my DAW setup again - maybe another month or so.

I'm thinking to compare to other multi-FX units which DO specify their dynamic range. The problem is: if you don't know how they define it then you'll be comparing apples to oranges. Fomr what I've seen on the HD500, dynamic range is "very good".

First you'll have to define dynamic range. Dynamic range can be defined as max usable input signal level - min input usable signal (a telecom/audio way of looking at things). Or it could be the difference between the max output signal and the max THD/spur/noise output components (a data converter way of looking at things, The biggest signal could be 0dBFS, or maybe the 10% THD point (which seems to be a standard in some audio applications) which is ideally is +2dBFS. I measured the 0dBFS point in this post here, which shows 8.3Vpp for the Guitar Input. Other inputs/outputs are also listed. The smallest signal is limited by the noise floor. Shall we use A-weighted noise or something else? Should the minimum signal be the noise floor itself or +3dB above the noise floor? Or some other level above the noise floor? 10-12dB is generally considered a good minimal SNR before humans consider a voice unusable. Does guitar have the same psycho-acoustic limit? All this to say, dynamic range is a "fuzzy" spec and not an industry standard. I'm not trying to make this more difficult than it is, but I need some guidance on what to give you. The max signal portion is already done in the thread listed above. The min signal portion has yet to be done, but is easy to measured with any DAW. But I mainly need to know if you want A-weighted noise or not. I'm thinking maybe yes. Or maybe there's another standard that should be used?

Yes, that issue is still with us. I use Guitar/Variax with my Gibson because 1) I don't use the Variax so it becomes a "not connected" input, 2) the noise (aka "hiss") level is minimized with this configuration, and 3) the Gibson output signal is hot enough that it NEEDS the 6dB input loss this configuration gives me. Otherwise I'd be clipping everything. It has worked very, very well for me.

I disagree. I learned a lot by looking thru custom tone designs. I didn't use ANY of them, but i learned a lot about how to do things by looking at them and playing with them.

Or maybe take those custom tone patches and remove the amp/cab. You're guitar amp already provides those. That may be enough to get you in the ballpark of what you're seeking.

The best place to start is to download some acoustic patches from custom tone. You may not use them but take a look at how they set things up. It'll help a lot. That being said, I use mine with acoustic instruments all the time and it's quite easy. No amp. Input 1 = guitar. Input 2 = Variax (i.e., unused). Then just plop in a compressor, one at a time, and see which one you like the best. You'll probably want to keep Drive fairly low to keep the grunge/crunch/dirt at bay. Then do the same for reverb. Try them one by one until you find the one you like. But be aware that what you like in your bedroom or headphones may NOT be what you like when playing live, Oh, you'll want the reverb to be the last thing in your signal path,

Sure it could. The headphone drivers are designed to drive very low impedance while the 1/4" outputs are not. Plugging low impedance headphones into the 1/4" outputs might cause a great deal of signal change.

Check out my discussion of IO gain here. I did not measure the noise output but I measured gain and clip levels of all the IO, including the headphone output. You can see the headphone output has about 15dB more gain than the 1/4" outputs. Clip levels are the same.

You could setup the tempo beforehand in a patch for that song and then invoke that patch. The LED will blink at the tempo rate. No noise. Just look at the LED and count it out.

You already have studio monitors so I assume you have a clean amp (as opposed to a guitar amp). If not, then get a clean amp. $200 won't get you a very good pair of studio monitors. They seem to start around $500/pr. Otherwise, if you can find an audio interface with SPDIF then get that.

Use? I have no idea. I just thought he wanted to see the notes. It sounded like the OP just wanted to check himself against a metronome in a DAW.

I would think you'd be able to pick out the start-of-notes if you: Use a very clean setting (i.e., no FX, no EQ, no amps) Pick only 1 note at a time (no chords, no double stops, clean arpeggios, etc) Staccato playing will be easiest to detect 120 bpm = 2 notes per second for each beat. You'll need to zoom in to see that time frame. The B on the E string 7th fret has ~500 periods each second (or 1 cycle every 2msec), or 250 periods per 1/4 note beat at 120 bpm. So that waveform should look sorta like a sinewave with 500 cycles of decaying amplitude until the next beat appears. You'll need to zoom in even more to see that level of detail in the waveform.

Try the 80A flanger with: Range = full counterclockwise (removes sweep) Enhance = 50% (or whatever you like) Manual = adjust to get the comb filter response you like (or set control for EXP1 to allow you to try different settings while you play) Odd/Even = select the one that sounds best to you.

I thought the biggest difference, besides the DSP horsepower, was the IO. The Pro has a few more options, doesn't it? That's because it's setup more for studio work while the HD500X is more for live work?

Parametric EQ: Set Freq = 10/11 for 98/102 Hz and set Freq = 23 for 201 Hz (you can't go all the way up to 100k/200kHz so I assume this is a typo). For -5dB set gain = 30%. Set Q for however wide you want it. High Q = very narrow. Low Q = wideband.

I believe the looper responds to MIDI controls. Perhaps get a MIDI foot switch.