[Live-devel] live audio source

[Stas Desyatnlkov]

Figured it. The live library uses preferred packet size. This size is
accommodated according to 1400 bytes - the best utilization of Ethernet
packet.
But in case of audio this is too much so I called:

//This takes care of 7 frames per packet problem and audio delay
sink->setPacketSizes(160*2 + 12, 160*2 + 12);



-----Original Message-----
From: live-devel-bounces at ns.live555.com
[mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson
Sent: Tuesday, March 14, 2006 8:32 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] live audio source


>I build the sample that sends uncompressed voice (8kHz, 1 chan, 16bits)
>over RTP (no RTCP and no RTSP). The sample runs on the local network.
>The problem is the voice delay is rather big 70 - 100 msec.

[...]

>I'd like to reduce voice delay down to 30-50msec range. What can be
done
>?

It's hard to say, because you can't do a good job of reducing delay 
until you know exactly where it's occurring.  There are many 
different potential sources of delay in a network-based application: 
capture, transmission, propogation delay, reception, 
rendering.  I.e., delay can occur at both ends, and in many 
places.  The "LIVE555 Streaming Media" code - even if it's used at 
both the sending and receiving ends - is only a small part of the 
overall system.

Before you can figure out how to reduce delay, you're going to have 
to figure out exactly where it is occurring.  I.e., you're going to 
have to dive in and instrument all parts of your system to figure this
out.


	Ross Finlayson
	Live Networks, Inc. (LIVE555.COM)
	<http://www.live555.com/>

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