[pulseaudio-discuss] [PATCH v6 19/25] loopback: Implement adaptive re-sampling

[Georg Chini]

On 23.08.2016 17:47, Tanu Kaskinen wrote:
> On Sun, 2016-06-05 at 21:05 +0200, Georg Chini wrote:
>> To account for the difference between source and sink time domains, adaptive
>> re-sampling is implemented. This massively improves latency stability. Details
>> of the controller implementation can be found in "rate_estimator.odt".
> I don't know whose definition is wrong, but to me "adaptive resampling"
> means just dynamically adjusting the resampling to make sure that there
> aren't buffer under- or overruns due to clock rate differences, and the
> existing code already satisfies this. Therefore, it looks wrong to me
> to say that this patch "implements adaptive resampling", as if that
> wasn't already implemented.
>
Up to this patch, it was always assumed, that source and sink are both
running with the nominal rate and there is only some difference between
target and actual latency that must be corrected. After the initial 
correction
you would see some saw-tooth function of the latency, because the
controller assumes that both sampling rates are equal and only corrects
the latency when it goes out of range. So the difference between the time
domains was not considered.
This patch now tries to find the optimum sampling rate for the sink, so
that the latency remains constant by accounting for the difference between
the time domains. If this is not adaptive re-sampling, how would you 
call it?