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 - 23:10:40 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.

The problem is (probably) the buffering in the application itself, and not
buffering in the network card or drivers. RAT and other lossy tolerant
network applications buffer the audio in order to remove jitter from the
stream of incoming UDP packets, as well as detect and reconstruct (or ignore)
lost data.

If you want to get below 50 ms, you will need an other application, which is
suited for LAN use, and not WAN (like rat). I don't know of one though.

/Anders Torger


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