06.02.2015 14:56, Georg Chini wrote:
> One more thing: There is a systematic error in the adjust_time I could
> not work around without
> introducing too much overhead. The latency snapshot varies widely in the
> execution time, I
> measured values between 50 us and more than 60 ms. So if the extreme
> values follow each other
> you will have one adjust time that is around 60 ms too long and another
> one which is 60 ms too
> short. Maybe this also contributes significantly to the (in)stability of
> regulation.
Well, I have looked into this issue.
Basically, you have a callback, time_callback, which is called every
u->adjust_time microseconds, according to a timer. All it does is to
send asynchronous messages to the sink input and the source output using
pa_asyncmsgq_send(), and they fill in various portions of the latency
snapshot by querying the relevant memblockq and the sink/source itself,
as well as snapshotting the total length of data received or sent.
A potential problem is that pa_source_get_latency_within_thread() and
pa_sink_get_latency_within_thread() themselves, in the case of an alsa
source, go through a smoother (see source_get_latency() in
src/modules/alsa/alsa-source.c), which _also_ tries to do sample rate
estimation for you! Try to avoid that, even though this means code
duplication.
Unfortunately, this is easy to recommend, but I can't really see how
this can be done.
The smoother is updated _after_ a successful write to the alsa device
(via traditional UNIX write or via mmap), while the pop callbacks are
executed just before that. So, they are called at the moment when the
influence from a bad rate estimation via the smoother is the greatest.
Now the suggestions.
I think that, ideally, for such use cases, it is important to have
timestamped latency snapshots for both sinks and sources in PulseAudio
core. This would mean introducing a new message that gets the latest
reliable latency snapshot (i.e., timestamp according to the wall clock,
send/receive counter, input/output buffer size, and the latency itself),
without any interpolation. If the sink does not implement this, just
fallback (in the generic sink code) to getting the current latency.
Also, because such snapshots for the sink and the source will not happen
at the same moment, you have to deal with it.
Also, I have a very heretic thought. Namely, that the smoother in the
alsa sink and source may actually be a bad idea and is better removed. I
have not tested this. But it is used only in two places: for reporting
latency (where it confuses your module) and for calculating the amount
of time to sleep in the case of timer-based scheduling (where even
module-alsa-sink does not trust the result, i.e. discards it if it is
greater than the non-transformed time interval). And, if I recollect
correctly, there were complaints about it being fooled by batch cards,
and they were cited as one of the reasons not to enable timer-based
scheduling on batch cards. So - maybe, for the purposes of timer
based-scheduling we should just assume the worst case, i.e. the card
that is, say, 0.75% faster than nominal, and use the nominal rate
together with the latest snapshot time in {source,sink}_get_latency()?
Basically, the fear is that the smoother makes a greater mistake in the
estimated rate than just assuming the nominal one. Maybe you can try
this suggestion?
For Tanu's patch status page: please leave the status of this patch as
unreviewed. The general idea of the patch does not look brilliant, but
it's the best known-working idea that we currently have on the topic,
and I have not reviewed all the fine details.
--
Alexander E. Patrakov
