On Wed, 2018-08-22 at 21:25 -0500, Pierre-Louis Bossart wrote:
> On 08/22/2018 07:46 PM, Guedes, Andre wrote:
> > On Tue, 2018-08-21 at 17:51 -0500, Pierre-Louis Bossart wrote:
> > > > > > +static int aaf_mclk_start_playback(snd_pcm_aaf_t *aaf)
> > > > > > +{
> > > > > > + int res;
> > > > > > + struct timespec now;
> > > > > > + struct itimerspec itspec;
> > > > > > + snd_pcm_ioplug_t *io = &aaf->io;
> > > > > > +
> > > > > > + res = clock_gettime(CLOCK_REF, &now);
> > > > > > + if (res < 0) {
> > > > > > + SNDERR("Failed to get time from clock");
> > > > > > + return -errno;
> > > > > > + }
> > > > > > +
> > > > > > + aaf->mclk_period = (NSEC_PER_SEC * aaf-
> > > > > > > frames_per_pkt) /
> > > > > >
> > > > > > io->rate;
> > > > >
> > > > > is this always an integer? If not, don't you have a
> > > > > systematic
> > > > > arithmetic error?
> > > >
> > > > NSEC_PER_SEC is 64-bit so I don't see an arithmetic error
> > > > during
> > > > calculation (e.g. integer overflow). Not sure this was your
> > > > concern,
> > > > though. Let me know otherwise.
> > >
> > > No, I was talking about the fractional part, e.g with 256 frames
> > > with
> > > 44.1kHz you have a period of 5804988.662131519274376 - so your
> > > math
> > > adds
> > > a truncation. same with 48khz, the fractional part is .333
> > >
> > > I burned a number of my remaining neurons chasing a <100 ppb
> > > error
> > > which
> > > led to underruns after 10 hours, so careful now with
> > > truncation...
> >
> > Thanks for clarifying.
> >
> > Yes, we can end up having a fractional period which is truncated.
> > Note
> > that both 'frames' and 'rate' are configured by the user. The user
> > should set 'frames' as multiple of 'rate' whenever possible to
> > avoid
> > inaccuracy.
>
> It's unlikely to happen. it's classic in audio that people want
> powers
> of two for fast filtering, and don't really care that the periods
> are
> fractional. If you cannot guarantee long-term operation without
> timing
> issues, you should add constraints to the frames and rates so that
> there
> is no surprise.
Fair enough. So for now I'll add a constraint on frames and rates to
unsure no surprises. Later we can revisit this and implement the
compesation mechanism you described below.
> >
> > From the plugin perspective, I'm not sure what we could do.
> > Truncating
> > might lead to underruns as you said, but I'm afraid that rounding
> > up
> > might lead to overruns, theoretically.
>
> Yes, you don't want to round-up either, you'd want to track when
> deviations become too high and compensate for it.
>
> >
> > > > > > +static int aaf_poll_revents(snd_pcm_ioplug_t *io, struct
> > > > > > pollfd
> > > > > > *pfd,
> > > > > > + unsigned int nfds, unsigned
> > > > > > short
> > > > > > *revents)
> > > > > > +{
> > > > > > + int res;
> > > > > > + snd_pcm_aaf_t *aaf = io->private_data;
> > > > > > +
> > > > > > + if (nfds != FD_COUNT_PLAYBACK)
> > > > > > + return -EINVAL;
> > > > > > +
> > > > > > + if (pfd[0].revents & POLLIN) {
> > > > > > + res = aaf_mclk_timeout_playback(aaf);
> > > > > > + if (res < 0)
> > > > > > + return res;
> > > > > > +
> > > > > > + *revents = POLLIN;
> > > > > > + }
> > > > >
> > > > > I couldn't figure out how you use playback events and your
> > > > > timer.
> > > >
> > > > 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.
- Andre
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