[wdmaudiodev] Re: RTAudio packet streaming is inefficient by design?

  • From: Robert Bielik <Robert.Bielik@xxxxxxxxx>
  • To: "wdmaudiodev@xxxxxxxxxxxxx" <wdmaudiodev@xxxxxxxxxxxxx>
  • Date: Thu, 21 Jun 2018 10:54:48 +0000

Hi Eugene,

But with small (units of ms) buffers, the time is not enough. For
example, with 10 ms buffer, you have only 5 ms to get the CPU, while
internal kernel delays may be long (see LatencyMon measurements).

Using at least four parts/packets, you might have 7.5 ms, which is more
reliable.

Indeed. But still, that only amounts to having a larger total latency, which 
would be the equivalent of having a 20 ms buffer (with 10 ms per buffer 
processing time). So I'm still not sure what the added benefit is. But I 
digress.

Anyway, do you, or anyone else, know how the case of one notification per 
buffer works ? Or even if it is used, at all ? I just can't grasp how things 
could possibly work with only one notification per buffer.

All the best
/Robert


Regards,
Eugene

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