All good things come to those who wait.. the Synth Squad 2 thread

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Amber (DCAM: Synth Squad) Cypher (DCAM: Synth Squad) Strobe (DCAM: Synth Squad)

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AMD FX8320@4.8ghz so i do get good performance out of it. Shall i post on the FX Forums?
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Please do!
This account is dormant, I am no longer employed by FXpansion / ROLI.

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Angus_FX wrote:@ musikmachine partly it's that Strobe2 runs at 2x oversampled by default (for the same or less cpu% than Strobe1 at 1x, on modern Intel chips). There are small improvements to the accuracy of the oscillator, filter & envelopes, but it's going from 1x to 2x that makes the biggest difference.

If you don't mind Strobe1's CPU load at 2x - with v2 you can go to 4x for the same amount (30% of one core @ 32 voices, 44.1K, i5-2500). It's designed to be playable in realtime all the way up to 8x on the recent generations of chips ("Sandy Bridge" aka i5/i7-2xxx and newer).

I wonder running it 2x oversampling at 192 Khz sample rate is equal to running it at 8x oversamplining at 48 Khz?
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I am more to Cypher. I like more complex possibilities. When it will be?
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chilly7 wrote:I wonder running it 2x oversampling at 192 Khz sample rate is equal to running it at 8x oversamplining at 48 Khz?
I'm nearly certain the answer is yes

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I have to say, Strobe 2 has become one of my most used synths, along with Serum. I love how the modulation system works, and Strobe 2 sounds amazing. I love this synth. :)
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@billcarroll thanks!


@chilly7
I wonder running it 2x oversampling at 192 Khz sample rate is equal to running it at 8x oversamplining at 48 Khz?
Almost. Control-rate (modulation update rate) is scaled as 32 samples from the base sample rate. So at 2x it's 64 (internal) samples, and at 8x it's 256 (internal) samples, but always 32 samples at the DAW's native sample rate.

So at 2x192 the effective control update rate is 6KHz, at 8x48 it's 1.5KHz. That'll mean somewhat higher CPU, but fractionally more accurate fast envelopes & high-rate LFOs. Unlikely to be audible though, most of the time.
I am more to Cypher. I like more complex possibilities. When it will be?
Expect some news on that in the next week or so. Work on it is progressing very nicely, I'll be able to say something about when, and how, it'll be available, very soon. It's safe to say it's not going to be available to the public in any form this side of New Year. There are some very early work-in-progress screenshots on our forum over at:

https://www.fxpansion.com/forum/viewforum.php?f=51

.. a certain amount has changed since those pics, I'll post an updated one along with all the other info as soon as I've got the OK from everyone here, probably second half of next week.
This account is dormant, I am no longer employed by FXpansion / ROLI.

Find me on LinkedIn or elsewhere if you need to get in touch.

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Cypher 2 looks excellent, much easier than the original. Great work!

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Angus_FX wrote:@billcarroll thanks!


@chilly7
I wonder running it 2x oversampling at 192 Khz sample rate is equal to running it at 8x oversamplining at 48 Khz?
Almost. Control-rate (modulation update rate) is scaled as 32 samples from the base sample rate. So at 2x it's 64 (internal) samples, and at 8x it's 256 (internal) samples, but always 32 samples at the DAW's native sample rate.

So at 2x192 the effective control update rate is 6KHz, at 8x48 it's 1.5KHz. That'll mean somewhat higher CPU, but fractionally more accurate fast envelopes & high-rate LFOs. Unlikely to be audible though, most of the time.
i am sorry.
1) What is effective control update rate?
2) What is control-rate (modulation update rate)

P.s.
I am Running Studio One Pro at 64 bit audio engine( Double percigion). (I am not toking about 64 bit OS or audio resolution)
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So in Strobe2, as in most softsynths and indeed a lot of analogue hardware synths (almost all polysynths) some signals (the audio path) run at full sample resolution, some things like LFOs, modulation etc. are updated at a lower rate, in Strobe2's case that's typically about 1.5KHz (DAW sample rate divided by 32), regardless of Strobe2's oversampling settings. Those calculations are quite expensive and not as aggressively optimised as the filter, oscillator, amplified code, so if you do them more often than is necessary for a smooth, accurate sound then you're just wasting CPU cycles.

If you run the DAW at a higher sample rate, it increases the control update rate. TBH, 32 samples is fast enough for /almost/ anything, so there's little advantage in doing so.
This account is dormant, I am no longer employed by FXpansion / ROLI.

Find me on LinkedIn or elsewhere if you need to get in touch.

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Angus_FX wrote:So in Strobe2, as in most softsynths and indeed a lot of analogue hardware synths (almost all polysynths) some signals (the audio path) run at full sample resolution, some things like LFOs, modulation etc. are updated at a lower rate, in Strobe2's case that's typically about 1.5KHz (DAW sample rate divided by 32), regardless of Strobe2's oversampling settings. Those calculations are quite expensive and not as aggressively optimised as the filter, oscillator, amplified code, so if you do them more often than is necessary for a smooth, accurate sound then you're just wasting CPU cycles.

If you run the DAW at a higher sample rate, it increases the control update rate. TBH, 32 samples is fast enough for /almost/ anything, so there's little advantage in doing so.
Thank you.
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Angus_FX wrote:@billcarroll thanks!


@chilly7
I wonder running it 2x oversampling at 192 Khz sample rate is equal to running it at 8x oversamplining at 48 Khz?
Almost. Control-rate (modulation update rate) is scaled as 32 samples from the base sample rate. So at 2x it's 64 (internal) samples, and at 8x it's 256 (internal) samples, but always 32 samples at the DAW's native sample rate.

So at 2x192 the effective control update rate is 6KHz, at 8x48 it's 1.5KHz. That'll mean somewhat higher CPU, but fractionally more accurate fast envelopes & high-rate LFOs. Unlikely to be audible though, most of the time.
May i ask another question. So if i record at 192 Khz/32bit sample rate at 64bit audio engine is there any benefit, even the smallest one running DCam Syth Squad more then 2x oversampling?
P.s. Maby for real time it will be hard but for rendering song i can set any oversampling.
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May i ask another question. So if i record at 192 Khz/32bit sample rate at 64bit audio engine is there any benefit, even the smallest one running DCam Syth Squad more then 2x oversampling?
Strobe and Amber? Probably not.

Cypher? If you go crazy with FM, waveshaping, drive etc. you can generate a lot of HF. So it's possible that it might be worth it, but only in very extreme cases.
This account is dormant, I am no longer employed by FXpansion / ROLI.

Find me on LinkedIn or elsewhere if you need to get in touch.

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Angus_FX wrote:
May i ask another question. So if i record at 192 Khz/32bit sample rate at 64bit audio engine is there any benefit, even the smallest one running DCam Syth Squad more then 2x oversampling?
Strobe and Amber? Probably not.

Cypher? If you go crazy with FM, waveshaping, drive etc. you can generate a lot of HF. So it's possible that it might be worth it, but only in very extreme cases.
Thank you.
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1) I noticed that LP4 filter in Dcam Synth Sqade fully open still eats alot of high frequancy. It would be nice if there were placed an option to extend frequancy range of filter.

2) Ability to increase polyphony to unlimited. Because in some cases you might need more poliphony then 42 (if i am not wrong).
For example 1) if you use unison of 12 then to get polyphony of 4 notes you need 48 polihony
2) If you want to get full decey of notes with long relise time. For example if i set long relise time and unison of 12 and i play cord of 4 notes 4 times i need polihpony of 48x4=192 There can be even more extreme cases for sauterne sounds.

3)It hard to explain with fwe words so. I set polyphony to 4. And i play 4 notes in the same time and then relise them and play another 4 notes. So the previouse 4 notes cut's off in order to play new 4 notes.
Right now:But when i play let sey 1 note and then relise that note and play a new note the previose note continue to decay because ther is still left unused polihony.
Want as option: But when i play let sey 1 note and then relise that note and play a new note the previose note cuts as if there is no polihony. Ofcause i do not want that always but is that possible to do as option?
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