Re: [linux-audio-dev] Broadcasting delays

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Subject: Re: [linux-audio-dev] Broadcasting delays
From: Anders Torger (torger_AT_ludd.luth.se)
Date: Mon Aug 13 2001 - 23:18:12 EEST


On Monday 13 August 2001 21:27, you wrote:
> Hi everybody !
>
> I´m most interested in audio broadcasting but still a newbie in Linux so
> there are a lot of questions :
> Because I´m a musician with the aim of making music with conferencing tools
> I should have a delay of < 50ms (at least first tests in the LAN).
>
> Former tests with WIN and SOLARIS gave me results of about 250 ms. I used
> RAT and later on I programmed my own tool with Java JMF which was even
> worst.
>
> After some postings in ALSA Mailinglist I was recommeded to move to this
> list.
> So can anyone tell me what to do to get a minimum delay. One year ago I
> heard of theoretical value of 25ms using RTP but I never could reach that.

Sorry for a second mail, but I forgot an important point: significant delay
is introduced in the coding of the audio as well, the sound is most often
compressed in 20 ms frames or similar, meaning that it will lag 20 ms even
before the audio has been sent to the network card buffers. However, since
the sound card sampling the incoming signal usually delays something similar
(or even more) due to block handling, this is most often ignored. The sound
card can usually be configured to deliver much smaller blocks though. The
same counts for the playback at the receiving end.

On a LAN, there should be enough bandwidth for uncompressed high quality
audio, and with some good coding it should certainly be possible to get below
50 ms, but I don't think there are tools for doing it out of the box.

/Anders Torger


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