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Cab Impulse Response Megathread

Does anyone have experience with loading Shift Lines free bass IR's into a Boss IR-2?

I've got an IR-2 on the way and plan on doing my own testing, but since an IR-2 with free bass IR's would be the cheapest way to get a good sound from your bass to a DAW, I think it would be great to at least write some of it down.
 
Anyone here use IRs that are not modeled after a cab? For example a pre set resonant HPF+LPF or an IR capture of a Noble DI signal? Anything out there?
No help on the IR capture part but this is close to my headphone jamming setup now after years of using IRs.

Noble + fairly aggresive LPF that also cuts some lower midrange, I find choosing my own EQ curve more fun at the moment than IRs
 
Hi all, I'm writing after I've read too many pages about IR. My head is burning. o_O
I'm used to play ampless with a nice GK Plex Preamp connecting it directly to the PA with DI with chorus and delay in send/return. In order to improve the sound sent to he PA now I would like to add a cab sim but I've a lot of doubts:
  1. where: I send the signal to PA by Plex Preamp. If the IR loader hasn't a DI, can I put it in send/return after the modulation effects?
  2. which: I'm trying a Mooer Radar of my guitarist but the sound easily clipped (now I'm powering by a 9v charger) and I would prefer some IR loader chargeable by 9v.
  3. how: power amp yes or power amp not? It seems with power amp ON the sound is better, it is sounds right to you?
  4. what: Darkglass Element is my wish but it's too expensive. Two notes Cab M is a good option for sure but I prefer something smaller. Now I'm intrigued by Nux Studio Mini and Tc Electronic Impulse but both are without power amp.
Can you help me to make a right choice?
 
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Hi all, I'm writing after I've read too many pages about IR. My head is burning. o_O
I'm used to play ampless with a nice GK Plex Preamp connecting it directly to the PA with DI with chorus and delay in send/return. In order to improve the sound sent to he PA now I would like to add a cab sim but I've a lot of doubts:
  1. where: I send the signal to PA by Plex Preamp. If the IR loader hasn't a DI, can I put it in send/return after the modulation effects?
  2. which: I'm trying a Mooer Radar of my guitarist but the sound easily clipped (now I'm powering by a 9v charger) and I would prefer some IR loader chargeable by 9v.
  3. how: power amp yes or power amp not? It seems with power amp ON the sound is better, it is sounds right to you?
  4. what: Darkglass Element is my wish but it's too expensive. Two notes Cab M is a good option for sure but I prefer something smaller. Now I'm intrigued by Nux Studio Mini and Tc Electronic Impulse but both are without power amp.
Can you help me to make a right choice?
I'll add a vote here for the Shiftline Cabzone X/Bass. You can load up to 10 cabinet IRs at a time (you can either use the included files or upload 3rd party cab IRs). There are a handful of tube poweramp emulators as well that you can use (or turn off). The poweramp emulation to my knowledge is set to whatever is pre-loaded on the device only but there were definitely a few that I liked, and a few could get hairy, as in "power amp tubes pushed".

All that with an XLR out, jack out and an unaffected jack out if you want to send the signal to an onstage rig, but don't want the IR sound through that set up.

Out of the IR loaders I tried I liked the sounds and form of this unit the best (Compared to Joyo Cab Box, Nu-X Solid Studio, Mooer Radar and Hotone OMNI IR), plus it was the easiest to connect to a laptop and upload different IRs to.

 
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I opened Pandoras box - the land of IR's. Ok I have Tonex and Tonex One and bought some IR's and stuff for that. I have been playing with modeling since the early 90's. Consider it modeling or not, but I had a Digitech RP-5 and that was my first foray into something trying to model a cab. TBH, it sucked horribly...ok fast forward to today and I while I love my Tonex, I still haven't setup the thing to model my own rig... anyway I like to tinker and do unconventional things just because that how I do things and you learn interesting lessons that way.

Anyway I got one of those Leo Jaymz /M-Wave/Erok IR BOX loaders to play with and see how it worked primarily with my Zoom MS 60-B and Ihave other use cases for this for recording too. So off I go downloading some free IR's based on some of the recommendations in this thread. I found some really good useable ones with Shiftline, Panda, Science Amplification, and a couple of other sources. Some were ok, some I didn't like at all, and a few I really liked. So I naorrowed down the ones I liked and was pretty happy with that. Shiftline overall was my favorite, especially the Mesa 215, SVT, and the 115. I test my IR's by going through the setlist of my band as we play several styles and I can see where one IR works better than another. I am setting this up as one of my gig rigs.

So, the tinkerer in me decided how can I manipulate these IR's. These are .WAV files right? I am Waveman, so why don't mess with with them in my DAW. I wanted to experiment with blending IR's. Cool. I made several ones based on two IR's playing with stuff just combining the two, write it the pedal and see what it sounds like. First iterations were cool, but character of the IR's were seeming to conflict a little, so adjust the mixer on the one you think is getting in the way. That worked pretty well! Then well, why not throw on a compressor and adjust the bass. That did stuff to it, some was good, but not what was looking for exactly. But still a valid thing to do. Anyway, my mind was spinning (still is spinning with possibilities here). I got the idea, why not take the signals for a couple or three IR's, split them in the same exact locations, and splice them back togther. That worked! While it may be a crude way to do, it's works. When analyzing the wav form, I just picked spots where it behaves differently, such as the front few ms where the big part of the wave is, then as it settled down, and then a little after that where becomes this long narrow waveform.

One of my favorites so far is where I took the first part of the Mesa 215, second part was from blended signal of the two, and the long tail was from SVT. I figure the front part of this is the basic character of the cab, but the second part and long tail effect resonance and stuff like that. Again, crude fun, and worth trying verses blending two complete signals :)
 
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Recently new to the world of IR's - I've noticed that my device is only capable of handling IR's in 44.1k format. Some IR's I'd like to try are only available in 48k.

Is it possible to down-sample from 48k to 44.1? I'm pretty sure I have the software to do it, but haven't tried it yet. Anyone here have ever tried this?
 
Recently new to the world of IR's - I've noticed that my device is only capable of handling IR's in 44.1k format. Some IR's I'd like to try are only available in 48k.

Is it possible to down-sample from 48k to 44.1? I'm pretty sure I have the software to do it, but haven't tried it yet. Anyone here have ever tried this?

AI says:

Yes, it is possible to down-sample 48 kHz Impulse Responses (IRs) to 44.1 kHz, and you should be able to do this with most standard audio editing software. This is a common process in audio production, and with modern, high-quality sample rate conversion (SRC) algorithms, the impact on sound quality is generally imperceptible to the human ear.

The Process and Why It Works
  • Sample Rate Conversion (SRC): Most digital audio workstations (DAWs) and audio editors (like Audacity, Logic Pro, Reaper, etc.) have built-in functions to convert the sample rate of an audio file. When you import a 48 kHz file into a 44.1 kHz project (or vice versa), the software will often automatically perform this conversion.
  • Aliasing Prevention: A crucial step in proper down-sampling is applying a digital low-pass filter before reducing the sample rate. This anti-aliasing filter removes any frequency information above the Nyquist frequency of the target sample rate (which is 22.05 kHz for 44.1 kHz) to prevent high-frequency components from being "folded" down into the audible range as unwanted artifacts. Modern audio software handles this filtering automatically and effectively.
  • Audible Difference: Both 44.1 kHz and 48 kHz sample rates can reproduce frequencies up to or beyond the limits of human hearing (around 20 kHz). The extra data in a 48 kHz file is for frequencies above this range, which you generally cannot hear anyway. Therefore, a well-executed down-sampling should sound virtually identical in practice.

Recommendations
  • Try It Out: Since you have the software, the best way is to simply try it. Down-sample an IR and load it into your device to see how it works.
  • Use Quality Software: Ensure you are using a reputable DAW or dedicated audio editor for the conversion. Consumer-grade software might use less sophisticated conversion algorithms, but most professional tools have excellent SRC.
  • Check Your Device's Manual: While conversion is possible, it's worth a quick check of your device's manual to ensure there aren't any specific requirements or potential limitations regarding the converted files (e.g., file type, bit depth, or length). Most devices will accept standard WAV format files.
 
AI says:

Yes, it is possible to down-sample 48 kHz Impulse Responses (IRs) to 44.1 kHz, and you should be able to do this with most standard audio editing software. This is a common process in audio production, and with modern, high-quality sample rate conversion (SRC) algorithms, the impact on sound quality is generally imperceptible to the human ear.

The Process and Why It Works
  • Sample Rate Conversion (SRC): Most digital audio workstations (DAWs) and audio editors (like Audacity, Logic Pro, Reaper, etc.) have built-in functions to convert the sample rate of an audio file. When you import a 48 kHz file into a 44.1 kHz project (or vice versa), the software will often automatically perform this conversion.
  • Aliasing Prevention: A crucial step in proper down-sampling is applying a digital low-pass filter before reducing the sample rate. This anti-aliasing filter removes any frequency information above the Nyquist frequency of the target sample rate (which is 22.05 kHz for 44.1 kHz) to prevent high-frequency components from being "folded" down into the audible range as unwanted artifacts. Modern audio software handles this filtering automatically and effectively.
  • Audible Difference: Both 44.1 kHz and 48 kHz sample rates can reproduce frequencies up to or beyond the limits of human hearing (around 20 kHz). The extra data in a 48 kHz file is for frequencies above this range, which you generally cannot hear anyway. Therefore, a well-executed down-sampling should sound virtually identical in practice.

Recommendations
  • Try It Out: Since you have the software, the best way is to simply try it. Down-sample an IR and load it into your device to see how it works.
  • Use Quality Software: Ensure you are using a reputable DAW or dedicated audio editor for the conversion. Consumer-grade software might use less sophisticated conversion algorithms, but most professional tools have excellent SRC.
  • Check Your Device's Manual: While conversion is possible, it's worth a quick check of your device's manual to ensure there aren't any specific requirements or potential limitations regarding the converted files (e.g., file type, bit depth, or length). Most devices will accept standard WAV format files.

Thanks, I may give it a go.

Just wanted to make sure I don't brick my device chasing down sampling rabbit holes (if that's possible) - I'll check the manual for any info regarding the specs of the converted file. :thumbsup:
 
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AI says:

Yes, it is possible to down-sample 48 kHz Impulse Responses (IRs) to 44.1 kHz, and you should be able to do this with most standard audio editing software. This is a common process in audio production, and with modern, high-quality sample rate conversion (SRC) algorithms, the impact on sound quality is generally imperceptible to the human ear.

The Process and Why It Works
  • Sample Rate Conversion (SRC): Most digital audio workstations (DAWs) and audio editors (like Audacity, Logic Pro, Reaper, etc.) have built-in functions to convert the sample rate of an audio file. When you import a 48 kHz file into a 44.1 kHz project (or vice versa), the software will often automatically perform this conversion.
  • Aliasing Prevention: A crucial step in proper down-sampling is applying a digital low-pass filter before reducing the sample rate. This anti-aliasing filter removes any frequency information above the Nyquist frequency of the target sample rate (which is 22.05 kHz for 44.1 kHz) to prevent high-frequency components from being "folded" down into the audible range as unwanted artifacts. Modern audio software handles this filtering automatically and effectively.
  • Audible Difference: Both 44.1 kHz and 48 kHz sample rates can reproduce frequencies up to or beyond the limits of human hearing (around 20 kHz). The extra data in a 48 kHz file is for frequencies above this range, which you generally cannot hear anyway. Therefore, a well-executed down-sampling should sound virtually identical in practice.

Recommendations
  • Try It Out: Since you have the software, the best way is to simply try it. Down-sample an IR and load it into your device to see how it works.
  • Use Quality Software: Ensure you are using a reputable DAW or dedicated audio editor for the conversion. Consumer-grade software might use less sophisticated conversion algorithms, but most professional tools have excellent SRC.
  • Check Your Device's Manual: While conversion is possible, it's worth a quick check of your device's manual to ensure there aren't any specific requirements or potential limitations regarding the converted files (e.g., file type, bit depth, or length). Most devices will accept standard WAV format files.


The bunk is in the details.

There's a bunch of nonsense that AI has scraped from internet discussions dealing in myths, misunderstanding, and good ole "makes sense to me..." Pseudoscience .

The reality is yes, it's a very straightforward, fundamental process.

Sample Rate Conversion works by massively up-sampling the audio, to a common integer between the source sample rate, and the destination sample rate, then truncates the redundant data.

Nothing is getting lost, or compromised - and certainly won't break any device you try to load the resultant file into.
 

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