[pulseaudio-discuss] alsa sink latency - how to account for startup delay

[Georg Chini]

Hi Tanu,

finally I figured it out where I went wrong. We are both right to some 
extent,
but (what is the important bit) you are right in so far, that the 
startup delay
should not be included in the latency reports. So sorry for the second mail
worm I have produced. As already mentioned in a previous mail we probably
could have sorted it out in half an hour if we had talked about it 
instead of
writing mails.
Let me explain what I mean:

First, please forget about pulseaudio for a moment, because we now talk
about a basic property of a stream which has nothing to do with the medium
that transports the stream.
Consider a source (just something producing samples, not the pulseaudio
source) and a corresponding sink. These are the preconditions:
- Source and sink share the same sampling rate, the time distance between
samples is dt = 1 / f.
- No samples will be dropped/inserted and no sample rate changes occur

At time t=0, the source produces the first sample.
There is some delay T, after which the first sample reaches the sink
and is played.
So the delay of the first sample is T.
Meanwhile, the source continues to produce samples.
At time t=dt the source produces the second sample.
Because source and sink share the same sampling rate, on the sink
side the second sample must be played at T + dt.
So the second sample has delay T + dt - dt = T.
At time t=2dt the source produces the next sample. On the sink side,
for the same reason as above, the third sample will be played at T + 2dt.
Again the delay is T.
And so on. How could this ever change without breaking the continuity
of the stream?

So you see, that the delay of the stream is set by the first sample and you
cannot change it, unless you use one of the stream manipulations I excluded
in the preconditions. This is where I am right.

On the other hand, in pulseaudio the number of samples in the sink is kept
constant until the sink really starts playing. This means, that the 
delay of the
top sample in the buffer will get smaller relative to "now" (though not 
relative
to the start of the first stream) until the startup delay has passed. 
Therefore
streams that are started after the startup delay has passed, will see only
the dynamic delay.
This is where you are right and why the startup delay should be ignored
in the latency reports. But still, from the perspective of the first stream,
this part is relevant as explained above and module-loopback has to get
rid of the excess latency (which it does).

I will change the code and the documentation accordingly which is pretty
simple. Again, sorry for being so slow to pick up the drift twice. I have to
admit that I am quite embarrassed. Thanks for your patience.

Regards
             Georg