[sdiy] My latest project

[cheater cheater]

What are you using the if branches for currently?

There are quite some techniques that can let you refactor your code,
but we would need to see the situation right now :)

D.

On Tue, Feb 16, 2010 at 19:00, Tom Wiltshire <tom at electricdruid.net> wrote:
>
> On 16 Feb 2010, at 16:59, Dave Manley wrote:
>
>> Tom Wiltshire wrote:
>>>
>>> Hi all,
>>>
>>> I'd like to show you what I've been up to recently:
>>>
>>> http://www.electricdruid.com/images/dualchannelvoice.jpg
>>>
>>> It's a digital/analog hybrid dual channel synth voice. The uP generates
>>> two audio channels (currently two pairs of PPG-style wavetable oscillators)
>>> and modulation sources (currently four LFOs - I haven't done any envelopes
>>> yet.) and outputs 8 CVs via four dual DACs. The 8 CVs control two SSM2044
>>> filters and dual stereo VCAs based on a SSM2164. So it's two channels of
>>> DualOsc/Filter/StereoVCA. It's also the thread that ties all the random
>>> questions I've been asking together.
>>> This evening I got the pan law for the stereo VCAs sorted out, and that's
>>> the last of the hardware working. It lives!
>>
>> Hi Tom,
>>
>> Looks like fun.  Can you give some details of how you implemented the
>> oscillator timing in the dsPIC?  Which features of the dsPIC are in use,
>> sample rate, etc?
>
> I had a conversation with Antti H. about it, and he gave me loads of good
> advice, like doing all the processing in blocks. I'd previously worked on a
> single sample at a time, but generating 16 at once is much faster. The
> sample rate is 89KHz, I think, blatantly not enough for the simple PPG
> oscillator scheme I'm using, but currently I don't care. That's a problem
> for later, if it's a problem at all. The audio DAC generates the sample rate
> clock itself, by requesting samples from the DMA hardware. Two DMA channels
> (L & R) are used to transfer data from the audio buffers to the DAC, so this
> doesn't tie up the processor. There are two buffers, and the DMA hardware
> generates an interrupt when it finishes a buffer. The interrupt sets a flag
> for the main code telling it which buffer is empty, and in the meantime the
> DMA reads the other buffer.
> The CV DACs are connected via SPI. Unfortunately the need to change chip
> select lines every two values makes using DMA impractical/ borderline
> useless for sending the CV data, otherwise I'd have used another DMA channel
> to move CV data from the final CV variables to the CV DACs. As it is, the
> main code has to do it, but it takes so little time to send 8 words at 10MHz
> that it isn't a problem.
> The modulation sources are calculated after the audio buffer is filled, so
> the modulation rate is 1/16th of the audio sample rate - over 5 KHz, plenty
> fast enough.
> I'm trying to apply my new DSP techniques to the modulation sources, and
> calculating them in a loop. As I said, I've done LFOs, all four calculated
> in one loop. I'm hoping to do something similar with envelopes. The tricky
> bit is finding ways to 'rephrase' things so that you don't have to have any
> code that branches - imagine writing an entire program with no "if"
> statements. Mostly it *can* be done, and I'm having great fun trying to
> squeeze more into less.
>
> T.
>
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