[NUT-devel] Another suggestion for broadcast [PATCH]

[Michael Niedermayer]

On Mon, Feb 11, 2008 at 11:04:17PM +0000, Måns Rullgård wrote:
> Michael Niedermayer <michaelni at gmx.at> writes:
> 
> > On Mon, Feb 11, 2008 at 12:14:20PM -0000, Måns Rullgård wrote:
> >> 
> >> Michael Niedermayer wrote:
> >> > On Thu, Feb 07, 2008 at 01:58:32PM -0000, Måns Rullgård wrote:
> >> >>
> >> >> Michael Niedermayer wrote:
> >> > [...]
> >> >> > That way the buffer_fullness stored in the syncpoint will always match
> >> >> > exactly the amount the decoder has in its buffer when reading the
> >> >> > syncpoint. If it has more in its buffer it just would change its clock
> >> >> > to one running at 100.1% and when it has less in its buffer it would
> >> >> > choose a 99.9% clock as reference. (Or any approximetaly equivalent
> >> >> > process)
> >> >>
> >> >> That the buffer fullness is off by N bits doesn't tell you how much too
> >> >> fast or too slow your clock is, only the sign of the error.
> >> >
> >> > yes
> >> >
> >> >
> >> >> Knowing
> >> >> also the magnitude of the error allows much more rapid convergence.
> >> >
> >> > I am not so sure about this. I mean i dont dispute that more
> >> > information should improve it, but i think its good enough with
> >> > the too much/too little.
> >> >
> >> > A simple example, lets assume we have a decoder with a clock that drifts
> >> > by up to D between syncpoints.
> >> > That is, in the most ideal case we would have to accept that we are
> >> > D off when we reach a syncpoint, assuming we synced to the previous
> >> > perfectly.
> >> >
> >> > Now lets assume that we are within -2D .. +2D at syncpoint x, and
> >> > we apply a +D correction if we are <0 and -D if we are >0. This
> >> > correction could be applied slowly until the next syncpoint. What
> >> > matters is that after the correction we are within -D .. +D and
> >> > with the drift thats again -2D .. +2D at syncpoint x+1.
> >> > Thus above is a proof by induction that just knowing the sign and
> >> > the worst case clock drift is sufficient to be within a factor of
> >> > 2 of the best achiveable clock sync. (comparing worst cases, not
> >> > average)
> >> 
> >> This is not how clock sync is usually done.  A typical implementation
> >> involves a PLL-type construct to make the local clock accurately track
> >> the sender clock.  Once locked, there is very little drift.  To correctly
> >> compensate for what little drift inevitably remains, the size of the
> >> error must be known.
> >
> > Could you elaborate on how PLL based clock sync with transmit ts works?
> > I am no PLL expert, what i know is more of the sort that a PLL takes a
> > reference signal like a sine wave as input, not occasional scalars which
> > represent time since some point 0.
> > Iam also fine with a RTFM + an url.
> 
> I recommend ISO 13818-1/ITU-T 222.0 Annex D.  It can be downloaded for
> free from http://www.itu.int/rec/T-REC-H.222.0-200605-I/en

thanks


> 
> >> The time difference can of course be computed from the difference in buffer
> >> fullness and the received bitrate, it merely takes a little more work on
> >> the receiver side.
> >
> > instead of transmit_ts one can use
> > internal_clock_ts + (buffer_fullness < real_fullness ? D : -D)
> > That should provide a pretty good reference for the PLL IMHO
> 
> This could easily end up oscillating around the correct rate, where
> using the precise error value would give a near-perfect match.

I assumed a normal clock which would be reasonably accurate, that is 1min/day
drift. In that case the oscilations would by <1ms per second with 1 
transmit_ts per second.
Though if one assumes a vastly inaccurate clock only stabilized by an input
voltage like described above the simple +-D method would work less well.
A simple improvment would be to double D if the error had the same sign as
last time and half it if it had opposit signs. This method should converge
within O(log N) time and stay pretty darn accurate once converged.



> 
> >> >> Providing the timestamp in the stream makes this trivial and independent
> >> >> of the buffering mechanism actually used.  Only specifying expected
> >> >> buffer fullness (according to a reference model) requires that the
> >> >> receiver at the very least simulate the reference model,
> >> >
> >> > I think most receivers will use something quite similar to the reference
> >> > model thus making this unneeded. Though yes a receiver using a different
> >> > buffer model might need to simulate the reference one.
> >> 
> >> I think it very unlikely that any real implementation will use whatever
> >> precise buffer model we choose.  Just about any implementation is
> >> likely to immediately extract the elementary streams of interest, and
> >> discard everything else, such as container headers and unwanted elementary
> >> streams.
> >
> > Well we can discard the container headers as well in the reference model.
> 
> We could, but I don't quite like the idea.  The difference in
> implementation effort is fairly small, and I prefer to keep the spec
> as simple as possible.

Iam fine with either ...


> 
> >> > But i have difficulty imageing a sufficiently different buffer model.
> >> > I mean a receiver with split buffers could just be taking the sum of
> >> > their buffer_fullness.
> >> > A reciver which removes packets later or not instantaneously would just
> >> > traverse the last few packets to find out how much the refernce buffer
> >> > would contain.
> >> 
> >> I'm not saying it would very difficult to simulate the reference buffer,
> >> but something is always more than nothing.
> >
> > yes but OTOH transmit_ts would not scale nicely with increased bandwidth.
> > Do you have some suggestions here to avoid this disadvantage?
> 
> Could you elaborate on this issue?

The issue i saw is impossible, i must have been tired or something :)

[...]
-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

Freedom in capitalist society always remains about the same as it was in
ancient Greek republics: Freedom for slave owners. -- Vladimir Lenin
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