SonicBirth v2 alpha

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Thanks it now works on yosemite! :D

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thank you sir!

I'm not sure if you want bug reports or not but:

Image

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Sure, send bug reports. Can't promise to reply, but I'll certainly look at them.

As for this bug, I did notice that a few days ago. It's on top of the todo list.

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Off topic: anyone in the US could order this:
http://www.amazon.com/Sesame-Street-Lea ... B002U6QQ9G
and ship it back to me (Montreal, Canada) ? I'll pay for the trouble and the shipping. PM me.

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Hi Makira,

It would be so nice to have custom GUI and "audio player" / "audio file" components on the new build.

Thanks for your work!

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Is there currently any way to set plug-in delay compensation? I'm making linear phase FIR filters and delay compensation is a must.

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Hi,

I have not tried v2 Alpha for some time, but I thought I should test making filters again.
So i tried making a first order lowpass filter to see how it behaved. It did not go well. At 44.1KHz sample freq. it would (again) behave as a first order at low frequencies, but above 15KHz or so, the level would start to increase. It does not make sense. I now found the problem. By bypassing the z-1 module, it started behaving just as it should.
I found that the adder module introduces a one sample delay! So in my lowpass filter i got two samples delay instead of one, when the z-1 module was used...
This was the problem I had last year... :?

Postby makira; Tue Nov 12, 2013 6:16 pm
RogerJoensson: What is the sample rate shown in the 'Sound & Midi' section of the UI ? It's normal to have issues when you go near/above half the sample rate. See sampling theorem. At the moment, SB will let you shoot yourself in the foot, it's up to you to limit parameters to sensible values.
Edit: Yesterday I did a test where I put one adder module in series with the signal from the sine osc. and added it with the direct signal. The output would go towards canceling out starting at about 1/4 fs. Today I can reproduce this test. But, the problem with the filter remains. It just works, when there is no delay at all in the circuit.
Here is the circuit. It will start to raise above 15KHz. Replace the z-1 with a wire and i acts as it should. Weird.
http://www.lineaudio.se/div/SonicB/firs ... lter.sb2au

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I believe this is because you're doing a feedback loop. In order to feed a signal back into itself it would make since that it'd require 1 sample delay. So sonicbirth automatically adds a sample of delay every time you use feedback. You can see exactly where the sample of delay occurs by the slight yellow highlight of the connection line. Notice the one coming off of the add component is yellowish. Yellow line = z-1.

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From the Sonicbirth website "Colored wires: grey is dead/inactive, black is live/active, and yellow is a feedback wire that adds one sample of delay."

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Correct. Yellow wires have a one sample delay. That was introduced sometime along the alphas. The other option was to prevent the wire from being added.

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dtpietrzak wrote:Is there currently any way to set plug-in delay compensation? I'm making linear phase FIR filters and delay compensation is a must.
Not currently, but I understand your need. It's on the todo list. A small update: I've hired someone to help with the custom widgets for the custom plugin's GUI. I've also hired another person to work on the Windows port. So while I don't have much time for Sonic Birth at the moment, it is still being worked on. I don't think I'll have time this month to release any new features, but most likely January there will be. Worst case I'll just update the build with an extended expiration date.

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makira wrote:Correct. Yellow wires have a one sample delay. That was introduced sometime along the alphas. The other option was to prevent the wire from being added.
OK. Thank you. I see. It is not very much yellow, though (more like grey with a brownish/greenish tint), but I do see the difference now.

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makira wrote:I've also hired another person to work on the Windows port.
:party: :party: :tu: :clap: :love: :party: :party: :harp: :band: :hail:

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Hi makira,

I discovered if a circuit has the "random" component in it, it will not compile as a Win VST.

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dtpietrzak wrote:Hi makira,

I discovered if a circuit has the "random" component in it, it will not compile as a Win VST.
never mind me. you're aware of this.

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