[LAD] [Jack-Devel] jack2's dbus name

Florian Schirmer florian.schirmer at native-instruments.de
Thu Jun 18 21:20:05 UTC 2009


On 18.06.2009, at 22:27, Jussi Laako wrote:

> No, generally data needs to be fed _to_ application immediately when  
> it
> becomes available after A/D conversion (PCI DMA completion interrupt).
> Application(s) process the data and it is ought to go to D/A  
> conversion
> on next hardware interrupt (PCI DMA reprogram interrupt), along with
> time-synchronous data from other applications. This creates total
> latency of inputhw+blocksize+outputhw. Generally input and output
> latencies should be around few tens of samples (due to delta-sigma
> converter resampling filters etc). And generally blocksize is also  
> kept
> around 64 or so.

I think this is incorrect. The total latency should be inputhw 

At timeoffset 0 the soundcard will start to capture samples. Until the  
first sample makes it into the buffer there is a delay of inputhw (A/ 
D, transfer etc). You can't access the buffer until at least blocksize  
samples have been captured. So the first buffer switch takes place at  
timeoffset inputhw + blocksize.

Now you do your processing. No matter how fast you can do that you'll  
have to wait until the next buffer switch will take place, which will  
happen at timeoffset inputhw + blocksize + blocksize.

Transfer + D/A will add outputhw to the mix and you end out with the  
first sample at your speakers at timeoffset inputhw + blocksize +  
blocksize + outputhw.


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