[Icecast] Delay on preroll

David Saunders abitar.com at gmail.com
Wed Dec 4 16:45:18 UTC 2019

One thing is, there is always a delay, especially if use a burst cache. I
can put 3 different clients on the delay can be a little annoying. From a
simple each or reverb effect to a couple sec delay. And if you separate on
remote network it can easily be 10 to 15 sec between clients.
 The delay is complicated mix of client's , network's and server's
configuration and speed.
 Big burst is good to get a client buffers filled so you get little
buffering when listing live.
  Twerking around the settings can make it better but can't be perfect.
  I find trying listen with the source in one ear and the stream with
other, you can find where this delay is. And it will change overtime.

On Wed, Dec 4, 2019, 6:52 AM Geoff Shang <geoff at quitelikely.com> wrote:

> On Wed, 4 Dec 2019, Javier Fuentes wrote:
> > I have a problem when I use the intro to send a pre-roll, the main stream
> > generates a 20 second delay, when the main stream is played it has no
> delay.
> > Is there any parameter I can use to eliminate the delay?
> > with pre-rolli:
> https://radiomarca-rrcast.flumotion.com/radiomarca/live.mp3
> > without pre-rolli:
> > https://radiomarca-rrcast.flumotion.com/radiomarca/live-np.mp3
> Are you saying that the listener is further behind on the stream with the
> pre-roll?
> I believe that this happens because the pre-roll is sent as quickly as
> possible by Icecast to the client, rather than at the rate of the audio.
> So it will take much less time to send than it does to play.  Once it has
> finished sending the pre-roll, it starts sending the live audio.  The
> player still has to play the pre-roll and will therefore be further behind
> the live audio when it gets to it.
> Note that I'm only guessing but it makes sense.
> My advice, use a shorter pre-roll.  As a listener, this will get very
> annoying very quickly.  40 seconds is a long time to listen through a
> pre-roll if your connection drops for a second.  I would limit it to 10
> seconds max.
> BTW: At time of writing, the live audio is out of phase (see
> https://www.audiocheck.net/audiotests_polaritycheck.php for explanations
> and examples).  this will cause the encoder to work much harder than it
> has to and will not sound as good as it should.  It will also cause anyone
> listening through a single speaker not to be able to hear the announcer
> and anything else that is panned centre (e.g. song vocals).
> this is most likely caused by an incorrectly wired cable or an RTS plug
> which is not fully inserted into the jack.
> HTH,
> Geoff.
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