[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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