Every time aaf->timer_fd expires, the audio buffer is consumed
by
the
plugin, making some room available on the buffer. So here a
POLLIN
event is returned so alsa-lib layer can copy more data into the
audio
buffer.
When there are two audio clock sources or timers that's
usually
where
the fun begins.
Regarding scenarios with two audio clock sources or timers, the
plugin
doesn't support them at the moment. This is something we should
work on
once the basic functionality is pushed upstream.
I was talking about adjusting the relationship between your
CLOCK_REALTIME timer and the media/network clock. I don't quite
get
how
this happens, I vaguely recall there should be a daemon which
tracks
the
difference between local and media/network clock, and I don't see
it
here.
Oh okay, I thought you were talking about something else :)
I believe you are referring to the gptp daemon from Openavnu [1].
The
AAF plugin doesn't use it. Instead, it uses linuxptp [2] which is
distributed by several Linux distros.
Linuxptp provides the phc2sys daemon that synchronizes both system
clock (i.e. CLOCK_REALTIME) and network clock (i.e. PTP clock). The
daemon disciplines the clocks instead of providing the time
difference
to applications. So we don't need to do any cross-timestamping at
the
plugin.
Humm, I don't get this. The CLOCK_REALTIME is based on the local
oscillator + NTP updates. And the network clock isn't necessarily
owned
by the transmitter, so how do you adjust?
When phc2sys is running, CLOCK_REALTIME is based on local oscillator +
phc2sys updates. The daemon keeps adjusting CLOCK_REALTIME based on PTP
clock via clock_adjtime syscall.
Is this a desirable "feature"? You may want to run other non-audio
applications where the NTP clock makes sense, e.g. to know when your
next call is, while playing audio on an external amp. Looks to me that
you've simplified the problem too much, or this is for custom audio-only
solutions. What am I missing?
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