Good morning Simon
yupp, and even the TURBO mode is supported, meaning your CC sequencing from the OT will be tighter
OT and the Heat will make a great combo, no doubt about it
Hmm, if I sell my Cali76TX I might be able to finance the 
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Is the envelope follower stereo so that the L and R channels can have different modulations based on the differences in the L/R signals, or is the signal to the EF section summed so that the two channels have one common EF? It’s not entirely clear from viewing the signal flow charts at page 5 in the manual.
The sonic state video was great. But what I’m missing is a video or some demos that showcase the filter properly. I’d really like to hear it in action…
The manual says nothing about unlinking the channels. Seems like it is only possible to process stereo signals.
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That is right… I am really hoping that it will come one day!
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I noticed the soundcard is not 96 hz, but 48 hz is this an issue or not.
Finally had a chance to read the manual fully. Seems like you can also use the heat as a de-esser and as a decent expander!
The only thing that IMO is missing is being able to unlink the channels for righteous M/S processing (and if I’m being greedy, 1 more parametric eq band), but even like this the heat is more tempting than I originally estimated.
With a little bit more fleshing out, this coulda been a good swiss-knife audio tool for any audio engineering situation! With easy DAW integration via OB, I feel this to be a missed opportunity somewhat. Perhaps some updates will be coming along in the future.
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Depends on the conversion quality AFAIK. “when in doubt 96k” approach is safer if the filtering on the ADC is compromized, but with a well engineered frontend and a nicely matching converter, 48k is totally game. 96k without a good clock is more prone to jitter anyway, right?
I look for setup ideas using the heat as a soundcard for mixing in ableton so you could prelisten songs via headphone. maybe splitted via prelisten via mac headphone outputs and master or channel heat to the main stage …
Now i just wish Elektron would create an app for tabled devices, mirroring ob setting to ipad, or just the small lcd screen bigger on mobile device for us elder people needing two differnt glasses, one for near and one for far .
Just noticed on the elektron site that it states “OB support coming soon”…is it not good to go as is and as was demonstrated in the sonicstate demo? If so, any estimation of when it will be OB ready?
I’ve seen no mention of this here and almost ordered, but then I just caught this small asterisk footnote. This is a concern for me as my intended use is 100% OB related!
Oh, if it’s just a Sierra issue no problem for me - happily using OB on Yosemite 
Still, it’s ambiguous as stated on the site. I’m sure Simon will clarify…
What do I need to use Overbridge?
You need a computer (Win/Mac OS) and DAW software. We recommend that the computer has Intel i-series processors or AMD equivalent. Windows Vista or lower is not supported, as well as Mac OS 10.6 and older.
So Yosemite is OK 
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actually 48 kHz is better for the workflow than 96 kHz. Your DAW Project wont use as many cpu resources at 44,1/48 kHz. 96 kHz really takes a lot of cpu power. Thats why I dont like to use the TR-8 in USB-Audio-Mode, because Roland only supports 96 kHz. From Sound quality, its really hard to tell a difference. 24 Bit is important, but 96 kHz mostly not.
Cheers,
Goat
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I would see only one potential interest for the 96 kHz : it should be interesting when one wants to timestretch the recording in the DAW afterwards, this should give less artefacts I guess, if I’m correct.
i guess not. The TS quality is dependent on the stretching algorhythm, not the sampling rate. But I didnt try that out personally, so Im not 100%sure. What I heard from Roland is, they say at 96 kHz virtual analog simulations are sounding better. but AH has real analog and just needs to make A/D and D/A conversions, so 44,1/48 kHz is absolutely fine, thats the rate I always mix at.
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Timestretching quality depends of the algorithm, of course, but I would thing that sampling rate (and bitrate) really make a difference for the same TS algo.
they should use oversampling internally anyway, so it shouldnt make a difference
Don’t know… you know, the basic timestretch algorithm basically change the pitch.
If you change the pitch down, you begin to see artefacts quite quickly.
I would say that it happens less quickly with 96kHz than with 48 kHz.
Like, one octave later.
But there are strong chances that I have no clue what I’m talking about !!
