Hi,
I often have this issue as well, which results in drops in the live stream
when a DJ is streaming to input.harbor.
Has anyone tried this `buffer.adaptative` yet? I'm trying to use it on my
input.harbor source, but I'm not sure what parameters I should set.
I have a setup kind of like this:
backup_playlist = playlist("./playlist.txt",conservative=true,mode=
"randomize",reload_mode="watch")
output.dummy(fallible=true,backup_playlist)
live_dj = input.harbor("live",port=9000)
live_dj = buffer.adaptative(live_dj,fallible=true)
on_fail = single("./technical_difficulties.wav")
source = fallback(track_sensitive=false,
[live_dj,backup_playlist,on_fail])
On Wed, Nov 5, 2014 at 2:50 PM, Ken Restivo <[email protected]> wrote:
>
> I'll have to try this adaptive buffer, thanks.
>
> I looked at it some more, with data from several DJ's scattered across
> North America.
>
> The DJ's streaming from New York (which is where the liquidsoap server is
> located), had uniformly perfect buffer statuses. In fact, one DJ even had a
> buffer whose size *increased* as his show went along!
>
> But the DJ's streaming from the west coast all uniformly had the same
> pattern of degradation: the buffer gradually decreasing over time.
>
> Now, unless sound cards run at different rates based on where they are
> physically located on the North American continent, I'm inclined to think
> the problem is one of network latency not of sound card sample rate.
>
> I have not taken inventory of which specific soundcards they are all
> using, but I'd guess it's the usual mix of SoundBlaster's, HDAIntels, and
> what not.
>
> I thought about perhaps there being some weirdness with power regulation
> and voltage: perhaps the east coast of the USA runs at 120VAC and the west
> coast at 115VAC or similar, or perhaps differences in the cycle frequency
> of the mains AC (60.1 Hz in east coast and 59.9Hz on west coast?) but the
> PC's power supplies, rectifiers, and voltage regulators should even all
> that out long before the soundcard gets its +5VDC powering its clock.
>
> Or am I missing something?
>
> -ken
> --
> --------
> On Mon, Oct 27, 2014 at 05:41:04AM -0700, David Baelde wrote:
> > Hi,
> > Depending on the DJʼs source client, the computerʼs clock can have an
> > impact too.
> > Anyway, if this is a serious issue you might want to look at an
> > operator that Samuel added not so long ago. Something like
> > buffer.adaptative iirc.
> > HTH
> > david
> > Brad Isbell a écrit:
> >
> > All sound cards run at slightly different rates. What you're looking
> > at is an encoder sending data at a slightly slower rate than you are
> > expecting. What is 44.1kHz to the DJ that is sending you data may
> > very well be 44.098kHz to the rest of the world.
> > The same problem occurs on the playback side of things. Some clients
> > will run faster, some will run slower. In practice, it doesn't
> matter
> > much. Most listeners don't listen long enough for there to be a
> > problem, and those that do will simply rebuffer for a second or two.
> >
> > [] Brad Isbell // AudioPump
> > [email protected]
> > Skype: bradisbell
> > Phone: +1 312-488-4680
> > On Fri, Oct 24, 2014 at 10:55 AM, Ken Restivo <[email protected]>
> wrote:
> >
> > Trying to wrap my brain around this data:
> > http://spaz.org/~ken/damfree2.pdf
> > That's a DJ connecting from a good DSL setup. The degradation is
> > very slow, and eventually hits a wall and stops, then starts again,
> > and then begins anew its descent.
> > I'm pretty sure their sample rate is set to 44.1khz (the same as the
> > server), but not entirely sure. Could it be sample rate mismatch,
> > but with the buffer taking much longer to empty than I've seen
> > before in sample-rate-mismatch cases? Or something else? Clocks out
> > of sync? It's baffling.
> > Thanks.
> > -ken
> > --------------------------------------------------------------------
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> > https://lists.sourceforge.net/lists/listinfo/savonet-users
> >
> > >
> >
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--
-Tony Miller
github.com/mcfiredrill
@freedrull
freedrool.us
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