[pulseaudio-discuss] [PATCH 07/13] loopback: Refactor latency initialization

Tanu Kaskinen tanuk at iki.fi
Thu Nov 26 10:43:21 PST 2015

On Thu, 2015-11-26 at 09:55 +0100, Georg Chini wrote:
> On 26.11.2015 08:41, Georg Chini wrote:
> > On 26.11.2015 01:49, Tanu Kaskinen wrote:
> > > On Wed, 2015-11-25 at 22:58 +0100, Georg Chini wrote:
> > > > On 25.11.2015 19:49, Tanu Kaskinen wrote:
> > > > > On Wed, 2015-11-25 at 16:05 +0100, Georg Chini wrote:
> > > > > > On 25.11.2015 09:00, Georg Chini wrote: I am not really sure where 
> > > > > > this discussion
> > is leading to. We are also mixing up different topics at the moment.
> > The first one is a matter of the safeguards. As already said in a
> > previous mail, in my opinion those safeguards only have to cover
> > the most common cases and do not need to be perfect because
> > the controller will take care at runtime.
> Let's pick up the safeguard discussion again. Maybe there is a reasoning,
> based on your calculations in the previous mails. As far as I can tell, we
> are talking about 3 different cases:
> 1) Interrupt driven alsa device
> I would propose to accept that for this kind of device the buffer_latency
> needs to be one default-fragment-size + some safety margin. This is a
> value proven in practice and when I remember correctly you reason in
> another mail that it could indeed fit in that special case.

Our calculations yield the same result in the usb sound card example
that we've been using, because your "average sink/source latency" is
effectively the same thing as the maximum sink buffer fill level, and
default-fragment-size in buffer_latency compensates for the missing
maximum source buffer fill level in the total latency calculation.

In my formula, the "baseline" for buffer_latency is zero. By that I
mean that if there weren't any complications like rate errors,
scheduling delays, latency measurement errors or "surprise latencies"
(latencies in sinks and sources that are higher than the configured
latency), then buffer_latency could be zero without any risk of

> 2) timer based alsa device
> I arrived at the conclusion that buffer_latency has to be 0.75 * 
> sink_latency.
> Would it be reasonable to argue that we have to keep one configured sink
> latency of audio around on the source side? If yes, it would make clear 
> where
> that 0.75 factor is coming from.
> The correct value for buffer_latency would then be
> buffer_latency = configured_sink_latency - 0.25 * configured_source_latency
> which gives 0.75 * configured_sink_latency if both are equal.

None of this makes sense to me, sorry. The starting point for
calculations is that the total latency has to be big enough to handle
the case where both the sink and the source buffers are full at the
same time. Once you have made sure that the total latency is at least
"configured source latency + configured sink latency", buffer_latency
doesn't have to be proportional to the sink latency.

> 3) the general case, so none of the above
> This is still unclear. As far as I understand, in this case there will never
> be timer based scheduling. Am I correct? If yes, it can be distinguished
> from case 1) just by checking the name of the device.
> In this general case we probably have to choose very conservative
> settings as we don't know exactly how the device will behave.
> If source and sink are of different device type we have to take the
> larger value.

I think all three cases can be handled using the same formula, when
"configured latency" is replaced with "maximum buffer fill level". If
my formula gives an overly conservative result in some case, that must
mean that the maximum sink or source buffer fill level is less than we
think. In case of timer-based scheduling in the alsa devices, I believe
it's correct to assume that the configured latency is the maximum
buffer fill level, modulo some safety margin that the alsa code uses
when configuring the wakeup timer.


More information about the pulseaudio-discuss mailing list