a technical question: what internal bit resolution is Cantabile working with? 24 bit integer of 32 bit float?
For me it is relevant for my gain staging: if all connections and gain sliders work at 32 bit float, there shouldnāt be problems with either clipping or leaving headroom before the final output. This means that I can
on on side leave ample headroom in my individual racks without losing bit resolution and simply adjust the overall output level before D/A conversion
on the other hand increase an audio inputās gain to a very hot level in order to drive a guitar amp plugin really hard without risking digital clipping (unless the amp itself is working at 24 bits fixed internal resolution)
if, OTOH, somewhere in Cantabileās routing, the signal path or processing were 24 bit fixed, Iād have to watch gain staging at every step in my setup (of course, I still need to do look at levels so that my plugins perform within reasonable parameters - thatās clear!)
So: is there a 24 bit fixed risk somewhere in Cantabileās signal path?
Cantabileās signal path is either 32-bit float or 64-bit audio if you enable it in Options ā Audio Engine.
Besides gain controls and mixing, the follow sample manipulations take place:
Conversion to sound card format - the very last thing and converts to the (often integer) format as required by the sound driver.
Conversion from sound card input - opposite of the above.
Using a 32-bit only plugin when 64-bit audio is enabled - the inputs to the plugin are narrowed to 32-bit and the outputs are widened to 64-bit.
Sample rate conversion (audio file or metronome sound at different sample rate to engine setting)
Audio file playback - format conversion from on-disk format to audio engine format.
Audio recording - format conversion from engine format to format selected for recordings (Options ā Recorder)
Output limiter - a tanh based limiter that affects signals above the limit selected in Options (as per above screen shot) - applied as the next last operation before output format conversion.
TL;DR - no, Cantabile doesnāt convert to integer/fixed within itās own audio pipeline.
So essentially I can proceed as intended - boosting signal levels to drive plugins as hot as I need, lower the level at the output stage of a rack to be sure I have enough headroom to mix everything together and then raise the output level at the end of the chain to maximize S/N ratio of my sound card.
One Q: can you say a bit more about the output limiter (found nothing in the Guides section)? if I select a threshold of 80%, does that mean that anything above 80% is flattened - or does it mean (my assumption) that the volume curve is linear up to 80% and then a gentle tanh(x) slope levels everything out up to 100%?
This would mean in musicianās term that the volume increment from 80% to 85% is still significant, whilst the increment from 95% to 100% (full scale) has a lot less impact (and any peak overshoot beyond 100% is simply flattened to 100%) - correct? So no harsh digital clipping should occur (within limits - driving a limiter too hard still sounds ugly ) even with the level meters hitting full scaleā¦
@brad: thinking about this for a couple of minutes - to be honest, I wouldnāt call this a LIMITER - itās actually more a SOFT CLIPPER or a saturator.
True limiting would mean that the overall volume of the total waveform is lowered as long as the threshold is exceeded, not just the peaks shaved off (albeit in a civilized way with the tanh mechanism).
But I guess that for most intents and purposes, this mechanism is enough to avoid any real nastiness from peaks in output; sometimes even a bit more useful than a true limiter - you can drive it hotter if you accept signal coloring.
Youāre absolutely rightā¦ in fact the functions I wrote to do this are called āFloat32SoftSatā and āFloat64SoftSatā. Definitely a soft saturation.
To be honest I canāt remember why I called the option āoutput limiterā - probably because I wanted something a bit more obvious in meaning.
And yes, the Wolfram stuff is very cool - first time Iāve used it. Usually I just use Googleās equation plotting, but couldnāt figure out how to do the piece-wise function. I like the Wolfram one better now.
I know this an old thread but I had a question. Do you still use the built in "limiterā or set up your own in a master rack? I ask because after I got āTrackLimitā from DMG and tried it on the master rack output for my setup and I feel that I get better fidelity. Also, if you do it this way what limiter do you like?
the built-in ālimiterā is more a āsoft clipperā, so its use is limited . It introduces distortion even when driven only mildly - which isnāt surprising in a soft-clipperā¦
I use compressors on both my main outputs rather than limiters. Most brick-wall VST limiters have quite significant latency as part of their forward-looking paradigm (e.g. Voxengo Elephant introduces 28ms of latency). While this doesnāt hurt in a mixing or mastering setup, it does hurt in a live playing scenario.
I use the fantastic FabFilter Pro-C - great-sounding, easy-to operate compressor. And whatever peaks slip through it will then be clipped by Cantabileās ālimiterā - I can live with thatā¦
Iām interested in your DMG TrackLimit, though - they claim that it is low-latency; how much latency does it truly introduce in your setup?
Thanks for the feed back on your output setup. I ran a check using Studio One resource utility and here are the numbers for Tracklimit. 128 sample buffer for audio.
There are six styles each with a different latency :
Latency
Aggressive - .2 mS
with ISP ON 1.2 mS
Punchy - .5 mS
with ISP ON 1.2 mS
Warm - 1 mS
with ISP ON 1.2 mS
Smooth - 2 mS
with ISP ON 2.2 mS
Tight - 3 mS
with ISP ON 3.2 mS
Transparent - 5 mS.
with ISP ON 5.1 mS
It employs a 2 stage internal arrangement which is supposed to protect transients. Real life I have been using it for 4 gigs and notice no lag running the punchy setting.
) even with the level meters hitting full scaleā¦
- takes me back to my computer science degree some 20+ years agoā¦