Another question about all of this. What if the user is using 24bit or higher in their A/D converter? Doesn't that make this all moot? Doesn't the added headroom prevent clipping and allow software limiters and compressors to work their magic without needing an extra hardware device at input? I'm still having a hard time adjusting to the idea that I'm going to have to be unlearning what I've learned over the last 20 years.audiojunkie wrote: ↑Tue Oct 15, 2019 4:10 pmFascinating information!! This has been very useful! I've lived under the false impression that a brickwall limiter would work just fine to prevent all clipping. It makes complete sense that clipping at the A/D converter input wouldn't work with a software limiter, but I guess I never even thought about it. I know that a brick wall limiter is great to prevent clipping in-track, but always assumed any clipping took place after the A/D converter. This discussion has corrected a 20+ year false impression! Very useful!!Anderton wrote: ↑Tue Oct 15, 2019 7:48 amAnything in the software is post-A/D converter, so it doesn't solve the problem of sending spiky transients into the A/D converter....which requires you to turn down the input level, and thus sacrifice input level. But there's much more to the story than that:audiojunkie wrote: ↑Mon Oct 14, 2019 11:27 pm So, what would be the difference between using a physical/hardware limiter prior to I/O hardware vs putting a software limiter (from within the I/O hardware--like a brickwall limiter vst)? Would it be intent or what one is trying to achieve?
* The LED has no overshoot, nobreathing, and nopumping. No active electronic circuit based on gain control can do that.
* The reaction time is instantaneous, so there's no need for look-ahead. The circuit lives in real time.
* The LED has junction capacitance, which increases with voltage. So the more level you put into it, the more capacitance there is (figure about 70 pF max, although that’s not an exact science). This not only shaves off the top, but does soft-, not hard-, clipping above the frequency of the guitar (i.e., where the spikes live).
* You can select an LED with a breakdown voltage to match your pickups. This is why the Gibson circuit worked so well - the pickups levels were a known entity. Luthier Jim DeCola went through a bunch of LEDs to find ones with the ideal breakdown voltage to shave off only the peaks. This is also why you have to use red LEDs. Other colors have too high a breakdown voltage.
Anyone who reads the Guitar Player article can see before-and-after waveforms, with and without the transient control circuit, as well as a chart showing the difference in average level that can go into an audio interface. The most dramatic difference was with the bridge pickup, which without the transient control circuit would peak out at 13.4V RMS before clipping the A/D converter. With the transient control circuit, it could peak out at 8.2V RMS. The neck and bridge pickups could deliver about 2 dB higher average level with the transient control circuit. It’s not as much, but still, 2 dB isn’t trivial.
It’s also not just about the A/D conversion. Eliminating that spike makes compressor and limiter gain control circuits soooo much happier. One of the people at Planet Waves who tested the cable said it made some amp sims sound better because they weren’t trying to deal with the spikes – similarly to how tubes “absorb” spikes.
Tom Scholz (the guitar player for Boston) is another red LED fan. We had quite a discussion about their distortion properties, when used instead of conventional silicon or germanium diodes in op amp feedback circuits. That was the secret of my hardware Quadrafuzz circuit (the one both Steinberg and MOTU virtualized).
I know, TMI...sorry. Back to the reviews.
To quote:
Note: Important part Bolded for easy reference.
https://www.tested.com/tech/1905-the-re ... bit-audio/
24-bit sound is a tricky thing to gauge. Does it provide for a greater resolution of sound? Definitively. It has room for 256 times the data, remember. Are you going to be able to hear that difference? Harder to judge. Human hearing supposedly tops out at 20kHz, but that doesn't make higher sample rates useless. According to the Nyquist rate, to fully capture a wave, it should be sampled at twice its highest frequency. In other words, a higher sample rate, and a greater bit depth, gives your sound more wiggle room, meaning sound peaks are less likely to be truncated and the subtleties of the music are less likely to be drowned out.